Thanx a lot! It is working now.

Here is the final dialplan for anyone paging cisco phone with PJSIP:

Code:

;Paging
exten => *71,1,Verbose(2,Paging device)
same  => n,Set(PageDevice=PJSIP/U_6995)
same  => n,Page(${PageDevice},ib(paging_handler^addheader^1))
same  => n,Hangup()


[paging_handler]
exten => addheader,1,Set(PJSIP_HEADER(add,Call-Info)=\;Answer-After=0)
same => n,Return()

Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101

---

You'll want to do:

Code:

same  => n,Page(${PageDevice},ib(paging_handler^addheader^1))

And not have a comma separating the i and b options.

Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101

---

Ok here is the final dialplan corrected (without the normal page before the pre-dialer)
It is still not working.

Code:

;Paging
;Paging
exten => *71,1,Verbose(2,Paging device)
same  => n,Set(PageDevice=PJSIP/U_6995)
same  => n,Page(${PageDevice},i,b(paging_handler^addheader^1))
same  => n,Hangup()

[paging_handler]
exten => addheader,1,Set(PJSIP_HEADER(add,Call-Info:\;Answer-After=0)=)

[Jun 12 08:34:00] -- Executing [*71@internal:1] Verbose("PJSIP/01-A-A4934CFE2736-0000199f", "2,Paging device") in new stack
[Jun 12 08:34:00] == Paging device
[Jun 12 08:34:00] -- Executing [*71@internal:2] Set("PJSIP/01-A-A4934CFE2736-0000199f", "PageDevice=PJSIP/U_6995") in new stack
[Jun 12 08:34:00] -- Executing [*71@internal:3] Page("PJSIP/01-A-A4934CFE2736-0000199f", "PJSIP/U_6995,i,b(paging_handler^addheader^1)") in new stack
[Jun 12 08:34:00] -- Called U_6995
[Jun 12 08:34:00] -- <PJSIP/01-A-A4934CFE2736-0000199f> Playing 'beep.ulaw' (language 'fr')
[Jun 12 08:34:00] == Using SIP RTP Audio TOS bits 184
[Jun 12 08:34:00] == Using SIP RTP Audio CoS mark 6
[Jun 12 08:34:00] -- PJSIP/U_6995-000019a0 is ringing
[Jun 12 08:34:00] <--- Transmitting SIP response (822 bytes) to UDP:10.188.6.24:5061 --->
[Jun 12 08:34:00] SIP/2.0 200 OK
[Jun 12 08:34:00] Via: SIP/2.0/UDP 10.188.6.24:5061;rport=5061;received=10.188.6.24;branch=z9hG4bK-46295178
[Jun 12 08:34:00] Call-ID: 6c1ef1bf-2e980314@10.188.6.24
[Jun 12 08:34:00] From: <sip:01-A-A4934CFE2736@XXX.XXX.XXX.XX>;tag=438399c348e9dae0o0
[Jun 12 08:34:00] To: <sip:*71@XXX.XXX.XXX.XX>;tag=68089e91-fc10-4405-ba80-fca0698656d1
[Jun 12 08:34:00] CSeq: 102 INVITE
[Jun 12 08:34:00] Server: Asterisk PBX 13.4.0
[Jun 12 08:34:00] Contact: <sip:XXX.XXX.XXX.XX:5060>
[Jun 12 08:34:00] Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
[Jun 12 08:34:00] Supported: 100rel, timer, replaces, norefersub
[Jun 12 08:34:00] Content-Type: application/sdp
[Jun 12 08:34:00] Content-Length: 237
[Jun 12 08:34:00]
[Jun 12 08:34:00] v=0
[Jun 12 08:34:00] o=- 25603912 25603914 IN IP4 XXX.XXX.XXX.XX
[Jun 12 08:34:00] s=Asterisk
[Jun 12 08:34:00] c=IN IP4 XXX.XXX.XXX.XX
[Jun 12 08:34:00] t=0 0
[Jun 12 08:34:00] m=audio 14664 RTP/AVP 0 101
[Jun 12 08:34:00] a=rtpmap:0 PCMU/8000
[Jun 12 08:34:00] a=rtpmap:101 telephone-event/8000
[Jun 12 08:34:00] a=fmtp:101 0-16
[Jun 12 08:34:00] a=ptime:20
[Jun 12 08:34:00] a=maxptime:150
[Jun 12 08:34:00] a=sendrecv
[Jun 12 08:34:00]
[Jun 12 08:34:00] <--- Received SIP request (693 bytes) from UDP:10.188.6.24:5061 --->
[Jun 12 08:34:00] ACK sip:XXX.XXX.XXX.XX:5060 SIP/2.0
[Jun 12 08:34:00] Via: SIP/2.0/UDP 10.188.6.24:5061;branch=z9hG4bK-e079fedc
[Jun 12 08:34:00] From: <sip:01-A-A4934CFE2736@XXX.XXX.XXX.XX>;tag=438399c348e9dae0o0
[Jun 12 08:34:00] To: <sip:*71@XXX.XXX.XXX.XX>;tag=68089e91-fc10-4405-ba80-fca0698656d1
[Jun 12 08:34:00] Call-ID: 6c1ef1bf-2e980314@10.188.6.24
[Jun 12 08:34:00] CSeq: 102 ACK
[Jun 12 08:34:00] Max-Forwards: 70
[Jun 12 08:34:00] Authorization: Digest username="01-A-A4934CFE2736",realm="XXX.XXX.XXX.XX",nonce="1434112440/42140018146b120209c0f890da7c9347",uri="sip:*71@XXX.XXX.XXX.XX",algorithm=MD5,response="e12fdd60191d101536a4bab7ff8012f0",opaque="315ef92c615f0d69",qop=auth,nc=00000001,cnonce="ee64ed90"
[Jun 12 08:34:00] Contact: <sip:01-A-A4934CFE2736@10.188.6.24:5061>
[Jun 12 08:34:00] User-Agent: Cisco/SPA514G-7.5.7
[Jun 12 08:34:00] Content-Length: 0
[Jun 12 08:34:00]
[Jun 12 08:34:00]
[Jun 12 08:34:00] > 0x7f82acc18bc0 -- Probation passed - setting RTP source address to 10.188.6.24:16462
[Jun 12 08:34:00] -- Channel PJSIP/01-A-A4934CFE2736-0000199f joined 'softmix' base-bridge <e54ff19a-9def-4da5-8ca6-430102694c26>
dti-asterisk*CLI>
Disconnected from Asterisk server
[Jun 12 08:34:02] Asterisk cleanly ending (0).
[Jun 12 08:34:02] Executing last minute cleanups

Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101

---

Your dialplan using a pre-dial handler has a normal Page followed by the Page with the pre-dial handler. It's unlikely that the Page with the pre-dial handler actually got invoked. What is the console output, and what is the SIP signaling (pjsip set logger on)?

Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101

---

Hi,

I had a working dialing code for paging with the Chansip channel. I'm having a hard time to get it to work with the PJSIP driver.

I had the following dialplan working in Chansip:

;Paging

Code:

exten => *71,1,Verbose(2,Paging)
same  => n,SIPAddHeader(Call-Info:\;answer-after=0)
same  => n,Set(PageDevice=SIP/U_6995)
same  => n,Page(${PageDevice},i)
same  => n,Hangup()

I have now this in PJSIP and the phone rings but it looks like the header is not added because the phone (Cisco) will not auto answer.

Code:

;Paging
exten => *71,1,Verbose(2,Paging device)
same  => n,Set(PageDevice=PJSIP/U_6995)
same  => n,Page(${PageDevice},i)
same  => n,Page(${PageDevice},i,b(paging_handler^addheader^1))
same  => n,Hangup()

[paging_handler]
exten => addheader,1,Set(PJSIP_HEADER(add,Call-Info:\;Answer-After=0)=)

I have also tried without the handler and I see no header added in wireshark.

Code:

;Paging
exten => *71,1,Verbose(2,Paging)
same  => n,Set(PJSIP_HEADER(add,Call-Info:\;answer-after=0)=)
same  => n,Set(PageDevice=PJSIP/U_6995)
same  => n,Page(${PageDevice},i)
same  => n,Hangup()

Statistics : Posted by phonefxg • on Fri Jun 12, 2015 5:53 am • Replies 7 • Views 101

---

There is no explicit mechanism to send a SIP NOTIFY to all SIP devices connected to either chan_sip or res_pjsip. Both provide a CLI command and AMI action to do it on an individual basis though. If you have a list you can call it for each device.

The CLI commands are:
sip notify
pjsip send notify

The AMI actions are:
SIPnotify
PJSIPNotify

Statistics : Posted by creationsphere • on Thu Jun 11, 2015 12:53 pm • Replies 1 • Views 90

---

Hi all, I am fairly new to Asterisk so please be gentle

I have been asked to set up Asterisk to send out a SIP Notify message once a day, to reboot attached ATA devices (its a temporary measure whilst a bug is fixed in the devices).

Here are my instructions...

Code:

About restart ATA by SIP NOTIFY message, when server send NOTIFY message which include header “Event: check-sync;reboot=true”, ATA will reboot:
For examples, SIP NOTIFY message:
NOTIFY sip:10002@192.168.0.35:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.251:5060;branch=z9hG4bK-d8754z-016ba529cb3ce414-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10002@192.168.0.251:5060>
To: <sip:10002@192.168.0.251:5060>
From: <sip:10002@192.168.0.251:5060>;tag=86598310
Call-ID: ZjYyODE1NDRiNjY2ZGI0YzcyMDk1YjdjMzZhOGEzY2I.
CSeq: 1 NOTIFY
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: Asterisk
Event: check-sync;reboot=true
Content-Length: 0

The trouble is I don't know how to do this! Asterisk is using realtime sip, in a sip_buddies database table. Each ATA has its own sip numbers (one for fxs and one for fxo). I have no idea how to get the ip address?

Please any help would be greatly appreciated!!

Statistics : Posted by creationsphere • on Thu Jun 11, 2015 12:53 pm • Replies 1 • Views 90

---

Hello,

Newbie here and I search for my question with no success so here goes.

I've got a trunk connection setup between my asterisk and a Cisco cme. There is a cisco phone registered to the CME and I've got a softphone (x-lite) registered to my asterisk

[cisco-phone]----(CME)====================(Asterisk)---------[x-lite]
ext:1000 ext:2000

I used the info in this page: https://learningnetwork.cisco.com/docs/DOC-3161 as a guide for my Asterisk config files.


!!!!!!!!!!!!!!!!!!!!!!! sip.conf !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
general
context = default

2000
type =friend
secret=1234
disallow=all
allow=ulaw

context=default

!!!!!!!!!!!!!!!!!!!!! extensions.conf !!!!!!!!!!!!!!!!!!!!!!!!!!!!


defaultexten => 2XXX,1,Dial(SIP/$) ; x.x.x.x Cisco Callmanager IP exten => 1XXX,1,Dial(SIP/$@x.x.x.x)


So from the Cisco phone I dial extension 2000 and I see from my xlite PC that extension 1000 is trying to reach me. I can answer the call and everything works fine.
The problem I have is when I dial ext 1000 from my xlite PC I get a error on my asterisk console basically telling me he doesn't know who extension 1000 is. And I'm assuming Asterisk is saying no one with extension 1000 is registered with me

So does anyone know how I can get my softphone to call the cisco phone on the other side of the trunk?

Statistics : Posted by panqa • on Wed Jun 10, 2015 7:18 pm • Replies 4 • Views 171

---

that is so big info, my eye is blind.

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

second part of sip debug log
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721085 ACK
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-HDVQ-1e622694-24c7e179
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (9 headers 0 lines) ---
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721085: Match Found
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 226721086 INVITE
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-IBRW-1e622696-571f33e0
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827251 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (13 headers 13 lines) ---
[Jun 13 00:31:15] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Found RTP audio format 0
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 8
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 101
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found audio description format PCMU for ID 0
[Jun 13 00:31:15] Found audio description format PCMA for ID 8
[Jun 13 00:31:15] Found audio description format telephone-event for ID 101
[Jun 13 00:31:15] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 100 Trying
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-IBRW-1e622696-571f33e0;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721086 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:15] Audio is at 19088
[Jun 13 00:31:15] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:15] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:15] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-IBRW-1e622696-571f33e0;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721086 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] Content-Length: 290
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=root 1445945467 1445945473 IN IP4 82.222.333.444
[Jun 13 00:31:15] s=Asterisk PBX 11.7.0
[Jun 13 00:31:15] c=IN IP4 82.222.333.444
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-16
[Jun 13 00:31:15] a=silenceSupp:off - - - -
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721086 ACK
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-GNBP-1e62269f-403beb19
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (9 headers 0 lines) ---
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721086: Match Found
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 226721087 INVITE
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-XDWY-1e6226a4-62ebb5de
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827253 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (13 headers 13 lines) ---
[Jun 13 00:31:15] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Found RTP audio format 0
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 8
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 101
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found audio description format PCMU for ID 0
[Jun 13 00:31:15] Found audio description format PCMA for ID 8
[Jun 13 00:31:15] Found audio description format telephone-event for ID 101
[Jun 13 00:31:15] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 100 Trying
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-XDWY-1e6226a4-62ebb5de;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721087 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:15] Audio is at 19088
[Jun 13 00:31:15] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:15] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:15] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-XDWY-1e6226a4-62ebb5de;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721087 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] Content-Length: 290
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=root 1445945467 1445945474 IN IP4 82.222.333.444
[Jun 13 00:31:15] s=Asterisk PBX 11.7.0
[Jun 13 00:31:15] c=IN IP4 82.222.333.444
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-16
[Jun 13 00:31:15] a=silenceSupp:off - - - -
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:15] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721087 ACK
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-WJAU-1e6226b2-6e958b1e
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (9 headers 0 lines) ---
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721087: Match Found
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 226721088 INVITE
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-YQNP-1e6226b4-73ae4d11
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827255 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (13 headers 13 lines) ---
[Jun 13 00:31:15] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Found RTP audio format 0
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 8
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 101
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found audio description format PCMU for ID 0
[Jun 13 00:31:15] Found audio description format PCMA for ID 8
[Jun 13 00:31:15] Found audio description format telephone-event for ID 101
[Jun 13 00:31:15] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 100 Trying
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-YQNP-1e6226b4-73ae4d11;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721088 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:15] Audio is at 19088
[Jun 13 00:31:15] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:15] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:15] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-YQNP-1e6226b4-73ae4d11;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721088 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] Content-Length: 290
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=root 1445945467 1445945475 IN IP4 82.222.333.444
[Jun 13 00:31:15] s=Asterisk PBX 11.7.0
[Jun 13 00:31:15] c=IN IP4 82.222.333.444
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-16
[Jun 13 00:31:15] a=silenceSupp:off - - - -
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:15] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:16] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721088 ACK
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] Max-Forwards: 30
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-DKZS-1e6226fd-3ba671d4
[Jun 13 00:31:16] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:16] Content-Length: 0
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------->
[Jun 13 00:31:16] --- (9 headers 0 lines) ---
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721088: Match Found
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:16] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:16] Content-Type: application/sdp
[Jun 13 00:31:16] CSeq: 226721089 INVITE
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] Max-Forwards: 30
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ELYR-1e6226fe-5a999079
[Jun 13 00:31:16] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:16] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:16] Content-Length: 265
[Jun 13 00:31:16]
[Jun 13 00:31:16] v=0
[Jun 13 00:31:16] o=cp10 143414827241 143414827257 IN IP4 10.7.1.154
[Jun 13 00:31:16] s=SIP Call
[Jun 13 00:31:16] c=IN IP4 91.121.129.146
[Jun 13 00:31:16] t=0 0
[Jun 13 00:31:16] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:16] b=AS:82
[Jun 13 00:31:16] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:16] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:16] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:16] a=fmtp:101 0-15
[Jun 13 00:31:16] a=ptime:20
[Jun 13 00:31:16] a=sendrecv
[Jun 13 00:31:16] <------------->
[Jun 13 00:31:16] --- (13 headers 13 lines) ---
[Jun 13 00:31:16] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] Found RTP audio format 0
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:16] Found RTP audio format 8
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:16] Found RTP audio format 101
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:16] Found audio description format PCMU for ID 0
[Jun 13 00:31:16] Found audio description format PCMA for ID 8
[Jun 13 00:31:16] Found audio description format telephone-event for ID 101
[Jun 13 00:31:16] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:16] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:16] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:16] SIP/2.0 100 Trying
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ELYR-1e6226fe-5a999079;received=91.111.222.33;rport=5060
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721089 INVITE
[Jun 13 00:31:16] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:16] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:16] Supported: replaces, timer
[Jun 13 00:31:16] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:16] Content-Length: 0
[Jun 13 00:31:16]
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------>
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:16] Audio is at 19088
[Jun 13 00:31:16] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:16] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:16] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:16] SIP/2.0 200 OK
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ELYR-1e6226fe-5a999079;received=91.111.222.33;rport=5060
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721089 INVITE
[Jun 13 00:31:16] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:16] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:16] Supported: replaces, timer
[Jun 13 00:31:16] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:16] Content-Type: application/sdp
[Jun 13 00:31:16] Content-Length: 290
[Jun 13 00:31:16]
[Jun 13 00:31:16] v=0
[Jun 13 00:31:16] o=root 1445945467 1445945476 IN IP4 82.222.333.444
[Jun 13 00:31:16] s=Asterisk PBX 11.7.0
[Jun 13 00:31:16] c=IN IP4 82.222.333.444
[Jun 13 00:31:16] t=0 0
[Jun 13 00:31:16] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:16] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:16] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:16] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:16] a=fmtp:101 0-16
[Jun 13 00:31:16] a=silenceSupp:off - - - -
[Jun 13 00:31:16] a=ptime:20
[Jun 13 00:31:16] a=sendrecv
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------>
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:16] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:16] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721089 ACK
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] Max-Forwards: 30
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-OZHG-1e622718-384bce3b
[Jun 13 00:31:16] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:16] Content-Length: 0
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------->
[Jun 13 00:31:16] --- (9 headers 0 lines) ---
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721089: Match Found
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:16] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:16] Content-Type: application/sdp
[Jun 13 00:31:16] CSeq: 226721090 INVITE
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] Max-Forwards: 30
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ZWHW-1e62271c-7e8d8127
[Jun 13 00:31:16] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:16] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:16] Content-Length: 265
[Jun 13 00:31:16]
[Jun 13 00:31:16] v=0
[Jun 13 00:31:16] o=cp10 143414827241 143414827259 IN IP4 10.7.1.154
[Jun 13 00:31:16] s=SIP Call
[Jun 13 00:31:16] c=IN IP4 91.121.129.146
[Jun 13 00:31:16] t=0 0
[Jun 13 00:31:16] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:16] b=AS:82
[Jun 13 00:31:16] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:16] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:16] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:16] a=fmtp:101 0-15
[Jun 13 00:31:16] a=ptime:20
[Jun 13 00:31:16] a=sendrecv
[Jun 13 00:31:16] <------------->
[Jun 13 00:31:16] --- (13 headers 13 lines) ---
[Jun 13 00:31:16] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] Found RTP audio format 0
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:16] Found RTP audio format 8
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:16] Found RTP audio format 101
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:16] Found audio description format PCMU for ID 0
[Jun 13 00:31:16] Found audio description format PCMA for ID 8
[Jun 13 00:31:16] Found audio description format telephone-event for ID 101
[Jun 13 00:31:16] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:16] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:16] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:16] SIP/2.0 100 Trying
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ZWHW-1e62271c-7e8d8127;received=91.111.222.33;rport=5060
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721090 INVITE
[Jun 13 00:31:16] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:16] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:16] Supported: replaces, timer
[Jun 13 00:31:16] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:16] Content-Length: 0
[Jun 13 00:31:16]
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------>
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:16] Audio is at 19088
[Jun 13 00:31:16] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:16] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:16] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:16] SIP/2.0 200 OK
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ZWHW-1e62271c-7e8d8127;received=91.111.222.33;rport=5060
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721090 INVITE
[Jun 13 00:31:16] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:16] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:16] Supported: replaces, timer
[Jun 13 00:31:16] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:16] Content-Type: application/sdp
[Jun 13 00:31:16] Content-Length: 290
[Jun 13 00:31:16]
[Jun 13 00:31:16] v=0
[Jun 13 00:31:16] o=root 1445945467 1445945477 IN IP4 82.222.333.444
[Jun 13 00:31:16] s=Asterisk PBX 11.7.0
[Jun 13 00:31:16] c=IN IP4 82.222.333.444
[Jun 13 00:31:16] t=0 0
[Jun 13 00:31:16] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:16] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:16] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:16] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:16] a=fmtp:101 0-16
[Jun 13 00:31:16] a=silenceSupp:off - - - -
[Jun 13 00:31:16] a=ptime:20
[Jun 13 00:31:16] a=sendrecv
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------>
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:16] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:16] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721090 ACK
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] Max-Forwards: 30
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-MSDW-1e62276d-7328844c
[Jun 13 00:31:16] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:16] Content-Length: 0
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------->
[Jun 13 00:31:16] --- (9 headers 0 lines) ---
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721090: Match Found
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:16] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:16] Content-Type: application/sdp
[Jun 13 00:31:16] CSeq: 226721091 INVITE
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] Max-Forwards: 30
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-CYBY-1e622843-1fd40f98
[Jun 13 00:31:16] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:16] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:16] Content-Length: 183
[Jun 13 00:31:16]
[Jun 13 00:31:16] v=0
[Jun 13 00:31:16] o=cp10 143414827241 143414827261 IN IP4 10.7.1.154
[Jun 13 00:31:16] s=SIP Call
[Jun 13 00:31:16] c=IN IP4 91.121.129.146
[Jun 13 00:31:16] t=0 0
[Jun 13 00:31:16] m=audio 33122 RTP/AVP 8
[Jun 13 00:31:16] b=AS:80
[Jun 13 00:31:16] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:16] a=ptime:20
[Jun 13 00:31:16] a=sendonly
[Jun 13 00:31:16] <------------->
[Jun 13 00:31:16] --- (13 headers 10 lines) ---
[Jun 13 00:31:16] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] Found RTP audio format 8
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:16] Found audio description format PCMA for ID 8
[Jun 13 00:31:16] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Jun 13 00:31:16] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:16] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw)
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:16] SIP/2.0 100 Trying
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-CYBY-1e622843-1fd40f98;received=91.111.222.33;rport=5060
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721091 INVITE
[Jun 13 00:31:16] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:16] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:16] Supported: replaces, timer
[Jun 13 00:31:16] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:16] Content-Length: 0
[Jun 13 00:31:16]
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------>
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (alaw) Video flag: True Text flag: True
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:16] Audio is at 19088
[Jun 13 00:31:16] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:16] SIP/2.0 200 OK
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-CYBY-1e622843-1fd40f98;received=91.111.222.33;rport=5060
[Jun 13 00:31:16] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721091 INVITE
[Jun 13 00:31:16] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:16] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:16] Supported: replaces, timer
[Jun 13 00:31:16] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:16] Content-Type: application/sdp
[Jun 13 00:31:16] Content-Length: 210
[Jun 13 00:31:16]
[Jun 13 00:31:16] v=0
[Jun 13 00:31:16] o=root 1445945467 1445945478 IN IP4 82.222.333.444
[Jun 13 00:31:16] s=Asterisk PBX 11.7.0
[Jun 13 00:31:16] c=IN IP4 82.222.333.444
[Jun 13 00:31:16] t=0 0
[Jun 13 00:31:16] m=audio 19088 RTP/AVP 8
[Jun 13 00:31:16] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:16] a=silenceSupp:off - - - -
[Jun 13 00:31:16] a=ptime:20
[Jun 13 00:31:16] a=recvonly
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------>
-- [Jun 13 00:31:16] -- Started music on hold, class 'default', on SIP/davidhp-00000006
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: channel.c:3577 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: channel.c:3740 ast_read_generator_actions: Generator got voice, switching to phase locked mode
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: channel.c:3577 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: channel.c:5361 set_format: Set channel SIP/davidhp-00000006 to write format slin
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: res_musiconhold.c:355 ast_moh_files_next: SIP/davidhp-00000006 Opened file 0 '/var/lib/asterisk/moh/nousVousTransferonsVersNotreSecretariat.2014'
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:16]
[Jun 13 00:31:16] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:16] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:16] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:16] CSeq: 226721091 ACK
[Jun 13 00:31:16] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:16] Max-Forwards: 30
[Jun 13 00:31:16] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:16] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-EIEU-1e622866-64fbfe86
[Jun 13 00:31:16] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:16] Content-Length: 0
[Jun 13 00:31:16]
[Jun 13 00:31:16] <------------->
[Jun 13 00:31:16] --- (9 headers 0 lines) ---
[Jun 13 00:31:16] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721091: Match Found
[Jun 13 00:31:16] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
......
[Jun 13 00:31:18] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:18] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:18] DEBUG[24926]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 199982f7475f10b60f537bde28f18e6b@127.0.1.1:15060 - OPTIONS (No RTP)
[Jun 13 00:31:18] DEBUG[24926]: chan_sip.c:3999 ast_sip_ouraddrfor: Target address 91.111.222.33:5060 is not local, substituting externaddr
[Jun 13 00:31:18] DEBUG[24926]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '199982f7475f10b60f537bde28f18e6b@127.0.1.1:15060' to '14f9a11e631b92ba1b8e9b9f04682364@82.222.333.444:15060'
[Jun 13 00:31:18] DEBUG[24926]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 14f9a11e631b92ba1b8e9b9f04682364@82.222.333.444:15060
[Jun 13 00:31:18] Reliably Transmitting (NAT) to 91.111.222.33:5060:
[Jun 13 00:31:18] OPTIONS sip:sip.ovh.fr SIP/2.0
[Jun 13 00:31:18] Via: SIP/2.0/UDP 82.222.333.444:15060;branch=z9hG4bK2d9982d6;rport
[Jun 13 00:31:18] Max-Forwards: 70
[Jun 13 00:31:18] From: "asterisk" <sip:0033123456789@82.222.333.444:15060>;tag=as5953578c
[Jun 13 00:31:18] To: <sip:sip.ovh.fr>
[Jun 13 00:31:18] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:18] Call-ID: 14f9a11e631b92ba1b8e9b9f04682364@82.222.333.444:15060
[Jun 13 00:31:18] CSeq: 102 OPTIONS
[Jun 13 00:31:18] User-Agent: Asterisk PBX 11.7.0
[Jun 13 00:31:18] Date: Fri, 12 Jun 2015 22:31:18 GMT
[Jun 13 00:31:18] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:18] Supported: replaces, timer
[Jun 13 00:31:18] Content-Length: 0
[Jun 13 00:31:18]
[Jun 13 00:31:18]
[Jun 13 00:31:18] ---
[Jun 13 00:31:18] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:18]
[Jun 13 00:31:18] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:18] SIP/2.0 501 Not Implemented
[Jun 13 00:31:18] Call-ID: 14f9a11e631b92ba1b8e9b9f04682364@82.222.333.444:15060
[Jun 13 00:31:18] CSeq: 102 OPTIONS
[Jun 13 00:31:18] From: "asterisk" <sip:0033123456789@82.222.333.444:15060>;tag=as5953578c
[Jun 13 00:31:18] To: <sip:sip.ovh.fr>;tag=00-28318-354713c3-0d25cb8f7
[Jun 13 00:31:18] Via: SIP/2.0/UDP 82.222.333.444:15060;received=82.222.333.444;rport=15060;branch=z9hG4bK2d9982d6
[Jun 13 00:31:18] Content-Length: 0
[Jun 13 00:31:18]
[Jun 13 00:31:18] <------------->
[Jun 13 00:31:18] --- (7 headers 0 lines) ---
[Jun 13 00:31:18] DEBUG[24926]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '14f9a11e631b92ba1b8e9b9f04682364@82.222.333.444:15060' of Request 102: Match Found
[Jun 13 00:31:18] Really destroying SIP dialog '14f9a11e631b92ba1b8e9b9f04682364@82.222.333.444:15060' Method: OPTIONS
[Jun 13 00:31:18] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
.......
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf00035e8' so dropping frame
[Jun 13 00:31:21]
[Jun 13 00:31:21] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:21] BYE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:21] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:21] CSeq: 226721092 BYE
[Jun 13 00:31:21] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:21] Max-Forwards: 30
[Jun 13 00:31:21] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:21] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:21] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-YOQB-1e6235aa-587becfb
[Jun 13 00:31:21] Reason: q.850;cause=31
[Jun 13 00:31:21] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:21] Content-Length: 0
[Jun 13 00:31:21]
[Jun 13 00:31:21] <------------->
[Jun 13 00:31:21] --- (11 headers 0 lines) ---
[Jun 13 00:31:21] DEBUG[24926][C-00000003]: chan_sip.c:26664 handle_request_bye: Initializing initreq for method BYE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:21] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:21] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:21] Scheduling destruction of SIP dialog '083fec101274a55057afb16861e87893@82.222.333.444:15060' in 6400 ms (Method: BYE)
[Jun 13 00:31:21]
[Jun 13 00:31:21] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:21] SIP/2.0 200 OK
[Jun 13 00:31:21] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-YOQB-1e6235aa-587becfb;received=91.111.222.33;rport=5060
[Jun 13 00:31:21] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:21] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:21] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:21] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:21] CSeq: 226721092 BYE
[Jun 13 00:31:21] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:21] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:21] Supported: replaces, timer
[Jun 13 00:31:21] Content-Length: 0
[Jun 13 00:31:21]
[Jun 13 00:31:21]
[Jun 13 00:31:21] <------------>
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: rtp_engine.c:1033 local_bridge_loop: rtp-engine-local-bridge: Ooh, got a hangup
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: channel.c:7992 ast_channel_bridge: Returning from native bridge, channels: SIP/davidhp-00000006, SIP/forfait-ovh-00000007
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: channel.c:2840 ast_hangup: Hanging up channel 'SIP/forfait-ovh-00000007'
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: chan_sip.c:7060 sip_hangup: Hangup call SIP/forfait-ovh-00000007, SIP callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: chan_sip.c:7065 sip_hangup: update_call_counter(0123456789) - decrement call limit counter on hangup
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: app_dial.c:3100 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: pbx.c:6573 __ast_pbx_run: Spawn extension (test,9,3) exited non-zero on 'SIP/davidhp-00000006'
== [Jun 13 00:31:21] == Spawn extension (test, 9, 3) exited non-zero on 'SIP/davidhp-00000006'
[Jun 13 00:31:21] DEBUG[25190][C-00000003]: channel.c:2661 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/davidhp-00000006'
-- [Jun 13 00:31:21] -- Stopped music on hold on SIP/davidhp-00000006
[

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

The device at 91.111.222.33 appears to be placing the call on hold using "sendonly" in an INVITE to Asterisk. Asterisk is doing as it is instructed.

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

with SIP debug:
an_sip.c:5736 do_setnat: Setting NAT on RTP to On
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:3642 obproxy_get: OBPROXY: Not applying OBproxy to this call
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:3999 ast_sip_ouraddrfor: Target address 91.111.222.33:5060 is not local, substituting externaddr
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:5736 do_setnat: Setting NAT on RTP to On
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '2e93a52f647f438e2577aff26eeb6108@127.0.1.1:15060' to '083fec101274a55057afb16861e87893@82.222.333.444:15060'
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: dsp.c:482 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: dsp.c:482 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPURI.
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:6356 sip_call: Outgoing Call for 0123456789
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (gsm|ulaw|alaw) Video flag: False Text flag: False
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:12] Audio is at 19088
[Jun 13 00:31:12] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:12] Adding codec 100002 (gsm) to SDP
[Jun 13 00:31:12] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:12] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:12] Reliably Transmitting (NAT) to 91.111.222.33:5060:
[Jun 13 00:31:12] INVITE sip:0123456789@sip.ovh.fr SIP/2.0
[Jun 13 00:31:12] Via: SIP/2.0/UDP 82.222.333.444:15060;branch=z9hG4bK05f7c36a;rport
[Jun 13 00:31:12] Max-Forwards: 70
[Jun 13 00:31:12] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:12] To: <sip:0123456789@sip.ovh.fr>
[Jun 13 00:31:12] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:12] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:12] CSeq: 102 INVITE
[Jun 13 00:31:12] User-Agent: Asterisk PBX 11.7.0
[Jun 13 00:31:12] Date: Fri, 12 Jun 2015 22:31:12 GMT
[Jun 13 00:31:12] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:12] Supported: replaces, timer
[Jun 13 00:31:12] Content-Type: application/sdp
[Jun 13 00:31:12] Content-Length: 313
[Jun 13 00:31:12]
[Jun 13 00:31:12] v=0
[Jun 13 00:31:12] o=root 1445945467 1445945467 IN IP4 82.222.333.444
[Jun 13 00:31:12] s=Asterisk PBX 11.7.0
[Jun 13 00:31:12] c=IN IP4 82.222.333.444
[Jun 13 00:31:12] t=0 0
[Jun 13 00:31:12] m=audio 19088 RTP/AVP 8 3 0 101
[Jun 13 00:31:12] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:12] a=rtpmap:3 GSM/8000
[Jun 13 00:31:12] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:12] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:12] a=fmtp:101 0-16
[Jun 13 00:31:12] a=silenceSupp:off - - - -
[Jun 13 00:31:12] a=ptime:20
[Jun 13 00:31:12] a=sendrecv
[Jun 13 00:31:12]
[Jun 13 00:31:12] ---
-- [Jun 13 00:31:12] -- Called SIP/0123456789@forfait-ovh
-- [Jun 13 00:31:12] -- Started music on hold, class 'default', on SIP/davidhp-00000006
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:3577 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:3740 ast_read_generator_actions: Generator got voice, switching to phase locked mode
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:3577 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: channel.c:5361 set_format: Set channel SIP/davidhp-00000006 to write format slin
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: res_musiconhold.c:355 ast_moh_files_next: SIP/davidhp-00000006 Opened file 0 '/var/lib/asterisk/moh/nousVousTransferonsVersNotreSecretariat.2014'
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2707 ast_rtp_write: Ooh, format changed from unknown to alaw
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:2742 ast_rtp_write: Created smoother: format: alaw ms: 20 len: 160
[Jun 13 00:31:12]
[Jun 13 00:31:12] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:12] SIP/2.0 100 Trying
[Jun 13 00:31:12] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:12] CSeq: 102 INVITE
[Jun 13 00:31:12] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:12] To: <sip:0123456789@sip.ovh.fr>
[Jun 13 00:31:12] Via: SIP/2.0/UDP 82.222.333.444:15060;received=82.222.333.444;rport=15060;branch=z9hG4bK05f7c36a
[Jun 13 00:31:12] Content-Length: 0
[Jun 13 00:31:12]
[Jun 13 00:31:12] <------------->
[Jun 13 00:31:12] --- (7 headers 0 lines) ---
[Jun 13 00:31:12] DEBUG[24926][C-00000003]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '083fec101274a55057afb16861e87893@82.222.333.444:15060' Request 102: Found
[Jun 13 00:31:12]
[Jun 13 00:31:12] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:12] SIP/2.0 407 authentication required
[Jun 13 00:31:12] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:12] Contact: <sip:0123456789@10.7.1.68:5060;user=phone>
[Jun 13 00:31:12] CSeq: 102 INVITE
[Jun 13 00:31:12] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:12] Proxy-Authenticate: Digest realm="sip.ovh.fr",nonce="0f1957835e74f2435fb7ec78144a8bb8",opaque="0f1952cd6be5262",stale=false,algorithm=MD5
[Jun 13 00:31:12] Record-Route: <sip:91.111.222.33:5060;transport=udp;lr>
[Jun 13 00:31:12] To: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b5-3664dbda1
[Jun 13 00:31:12] Via: SIP/2.0/UDP 82.222.333.444:15060;received=82.222.333.444;rport=15060;branch=z9hG4bK05f7c36a
[Jun 13 00:31:12] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:12] Server: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:12] Content-Length: 0
[Jun 13 00:31:12]
[Jun 13 00:31:12] <------------->
[Jun 13 00:31:12] --- (12 headers 0 lines) ---
[Jun 13 00:31:12] DEBUG[24926][C-00000003]: chan_sip.c:4529 __sip_ack: Acked pending invite 102
[Jun 13 00:31:12] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Request 102: Match Found
[Jun 13 00:31:12] Transmitting (NAT) to 91.111.222.33:5060:
[Jun 13 00:31:12] ACK sip:0123456789@sip.ovh.fr SIP/2.0
[Jun 13 00:31:12] Via: SIP/2.0/UDP 82.222.333.444:15060;branch=z9hG4bK05f7c36a;rport
[Jun 13 00:31:12] Max-Forwards: 70
[Jun 13 00:31:12] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:12] To: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b5-3664dbda1
[Jun 13 00:31:12] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:12] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:12] CSeq: 102 ACK
[Jun 13 00:31:12] User-Agent: Asterisk PBX 11.7.0
[Jun 13 00:31:12] Content-Length: 0
[Jun 13 00:31:12]
[Jun 13 00:31:12]
[Jun 13 00:31:12] ---
[Jun 13 00:31:12] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (gsm|ulaw|alaw) Video flag: False Text flag: False
[Jun 13 00:31:12] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:12] Audio is at 19088
[Jun 13 00:31:12] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:12] Adding codec 100002 (gsm) to SDP
[Jun 13 00:31:12] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:12] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:12] Reliably Transmitting (NAT) to 91.111.222.33:5060:
[Jun 13 00:31:12] INVITE sip:0123456789@sip.ovh.fr SIP/2.0
[Jun 13 00:31:12] Via: SIP/2.0/UDP 82.222.333.444:15060;branch=z9hG4bK6d10d631;rport
[Jun 13 00:31:12] Max-Forwards: 70
[Jun 13 00:31:12] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:12] To: <sip:0123456789@sip.ovh.fr>
[Jun 13 00:31:12] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:12] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:12] CSeq: 103 INVITE
[Jun 13 00:31:12] User-Agent: Asterisk PBX 11.7.0
[Jun 13 00:31:12] Proxy-Authorization: Digest username="0033123456789", realm="sip.ovh.fr", algorithm=MD5, uri="sip:0123456789@sip.ovh.fr", nonce="0f1957835e74f2435fb7ec78144a8bb8", response="03bf5f60fdaea739b4dde11387f0f549", opaque="0f1952cd6be5262"
[Jun 13 00:31:12] Date: Fri, 12 Jun 2015 22:31:12 GMT
[Jun 13 00:31:12] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:12] Supported: replaces, timer
[Jun 13 00:31:12] Content-Type: application/sdp
[Jun 13 00:31:12] Content-Length: 313
[Jun 13 00:31:12]
[Jun 13 00:31:12] v=0
[Jun 13 00:31:12] o=root 1445945467 1445945468 IN IP4 82.222.333.444
[Jun 13 00:31:12] s=Asterisk PBX 11.7.0
[Jun 13 00:31:12] c=IN IP4 82.222.333.444
[Jun 13 00:31:12] t=0 0
[Jun 13 00:31:12] m=audio 19088 RTP/AVP 8 3 0 101
[Jun 13 00:31:12] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:12] a=rtpmap:3 GSM/8000
[Jun 13 00:31:12] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:12] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:12] a=fmtp:101 0-16
[Jun 13 00:31:12] a=silenceSupp:off - - - -
[Jun 13 00:31:12] a=ptime:20
[Jun 13 00:31:12] a=sendrecv
[Jun 13 00:31:12]
[Jun 13 00:31:12] ---
[Jun 13 00:31:12] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 80 bytes
[Jun 13 00:31:12]
[Jun 13 00:31:12] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:12] SIP/2.0 100 Trying
[Jun 13 00:31:12] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:12] CSeq: 103 INVITE
[Jun 13 00:31:12] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:12] To: <sip:0123456789@sip.ovh.fr>
[Jun 13 00:31:12] Via: SIP/2.0/UDP 82.222.333.444:15060;received=82.222.333.444;rport=15060;branch=z9hG4bK6d10d631
[Jun 13 00:31:12] Content-Length: 0
[Jun 13 00:31:12]
[Jun 13 00:31:12] <------------->
[Jun 13 00:31:12] --- (7 headers 0 lines) ---
[Jun 13 00:31:12] DEBUG[24926][C-00000003]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '083fec101274a55057afb16861e87893@82.222.333.444:15060' Request 103: Found
[Jun 13 00:31:13]
[Jun 13 00:31:13] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:13] SIP/2.0 180 Ringing
[Jun 13 00:31:13] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:13] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:13] Content-Type: application/sdp
[Jun 13 00:31:13] CSeq: 103 INVITE
[Jun 13 00:31:13] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:13] Record-Route: <sip:91.111.222.33:5060;transport=udp;lr>
[Jun 13 00:31:13] To: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:13] Via: SIP/2.0/UDP 82.222.333.444:15060;received=82.222.333.444;rport=15060;branch=z9hG4bK6d10d631
[Jun 13 00:31:13] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:13] Server: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:13] Content-Length: 265
[Jun 13 00:31:13]
[Jun 13 00:31:13] v=0
[Jun 13 00:31:13] o=cp10 143414827241 143414827242 IN IP4 10.7.1.154
[Jun 13 00:31:13] s=SIP Call
[Jun 13 00:31:13] c=IN IP4 91.121.129.146
[Jun 13 00:31:13] t=0 0
[Jun 13 00:31:13] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:13] b=AS:82
[Jun 13 00:31:13] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:13] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:13] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:13] a=fmtp:101 0-15
[Jun 13 00:31:13] a=ptime:20
[Jun 13 00:31:13] a=sendrecv
[Jun 13 00:31:13] <------------->
[Jun 13 00:31:13] --- (12 headers 13 lines) ---
[Jun 13 00:31:13] DEBUG[24926][C-00000003]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '083fec101274a55057afb16861e87893@82.222.333.444:15060' Request 103: Found
[Jun 13 00:31:13] list_route: hop: <sip:91.111.222.33:5060;transport=udp;lr>
[Jun 13 00:31:13] Found RTP audio format 0
[Jun 13 00:31:13] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dc910
[Jun 13 00:31:13] Found RTP audio format 8
[Jun 13 00:31:13] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dc910
[Jun 13 00:31:13] Found RTP audio format 101
[Jun 13 00:31:13] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dc910
[Jun 13 00:31:13] Found audio description format PCMU for ID 0
[Jun 13 00:31:13] Found audio description format PCMA for ID 8
[Jun 13 00:31:13] Found audio description format telephone-event for ID 101
[Jun 13 00:31:13] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:13] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:13] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:13] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:13] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:13] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
-- [Jun 13 00:31:13] -- SIP/forfait-ovh-00000007 is ringing
-- [Jun 13 00:31:13] -- SIP/forfait-ovh-00000007 is making progress passing it to SIP/davidhp-00000006
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 103 INVITE
[Jun 13 00:31:15] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;transport=udp;lr>
[Jun 13 00:31:15] To: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Via: SIP/2.0/UDP 82.222.333.444:15060;received=82.222.333.444;rport=15060;branch=z9hG4bK6d10d631
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] Server: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827242 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (12 headers 13 lines) ---
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4529 __sip_ack: Acked pending invite 103
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Request 103: Match Found
[Jun 13 00:31:15] list_route: hop: <sip:91.111.222.33:5060;transport=udp;lr>
[Jun 13 00:31:15] set_destination: Parsing <sip:91.111.222.33:5060;transport=udp;lr> for address/port to send to
[Jun 13 00:31:15] set_destination: set destination to 91.111.222.33:5060
[Jun 13 00:31:15] Transmitting (NAT) to 91.111.222.33:5060:
[Jun 13 00:31:15] ACK sip:10.7.1.68:5060 SIP/2.0
[Jun 13 00:31:15] Via: SIP/2.0/UDP 82.222.333.444:15060;branch=z9hG4bK64912c68;rport
[Jun 13 00:31:15] Route: <sip:91.111.222.33:5060;transport=udp;lr>
[Jun 13 00:31:15] Max-Forwards: 70
[Jun 13 00:31:15] From: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] To: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 103 ACK
[Jun 13 00:31:15] User-Agent: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] ---
-- [Jun 13 00:31:15] -- SIP/forfait-ovh-00000007 answered SIP/davidhp-00000006
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: rtp_engine.c:1805 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/davidhp-00000006' with that of 'SIP/forfait-ovh-00000007'
-- [Jun 13 00:31:15] -- Stopped music on hold on SIP/davidhp-00000006
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: channel.c:5361 set_format: Set channel SIP/davidhp-00000006 to write format alaw
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: channel.c:3577 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: features.c:4250 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/forfait-ovh-00000007 since we're bridging
-- [Jun 13 00:31:15] -- Locally bridging SIP/davidhp-00000006 and SIP/forfait-ovh-00000007
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 226721082 INVITE
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-KNLG-1e6225bb-3a05f95e
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827243 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (13 headers 13 lines) ---
[Jun 13 00:31:15] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Found RTP audio format 0
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 8
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 101
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found audio description format PCMU for ID 0
[Jun 13 00:31:15] Found audio description format PCMA for ID 8
[Jun 13 00:31:15] Found audio description format telephone-event for ID 101
[Jun 13 00:31:15] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 100 Trying
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-KNLG-1e6225bb-3a05f95e;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721082 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:15] Audio is at 19088
[Jun 13 00:31:15] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:15] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:15] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-KNLG-1e6225bb-3a05f95e;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721082 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] Content-Length: 290
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=root 1445945467 1445945469 IN IP4 82.222.333.444
[Jun 13 00:31:15] s=Asterisk PBX 11.7.0
[Jun 13 00:31:15] c=IN IP4 82.222.333.444
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-16
[Jun 13 00:31:15] a=silenceSupp:off - - - -
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:15] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721082 ACK
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-HLRV-1e6225cd-54fa79a8
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (9 headers 0 lines) ---
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721082: Match Found
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 226721083 INVITE
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-GNFK-1e6225cf-3d4a273d
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827245 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (13 headers 13 lines) ---
[Jun 13 00:31:15] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Found RTP audio format 0
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 8
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 101
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found audio description format PCMU for ID 0
[Jun 13 00:31:15] Found audio description format PCMA for ID 8
[Jun 13 00:31:15] Found audio description format telephone-event for ID 101
[Jun 13 00:31:15] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 100 Trying
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-GNFK-1e6225cf-3d4a273d;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721083 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:15] Audio is at 19088
[Jun 13 00:31:15] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:15] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:15] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-GNFK-1e6225cf-3d4a273d;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721083 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] Content-Length: 290
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=root 1445945467 1445945470 IN IP4 82.222.333.444
[Jun 13 00:31:15] s=Asterisk PBX 11.7.0
[Jun 13 00:31:15] c=IN IP4 82.222.333.444
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-16
[Jun 13 00:31:15] a=silenceSupp:off - - - -
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:15] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721083 ACK
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-JFBF-1e62261a-298170d5
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (9 headers 0 lines) ---
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721083: Match Found
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 226721084 INVITE
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-PTSJ-1e62261c-21334a3c
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827247 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (13 headers 13 lines) ---
[Jun 13 00:31:15] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Found RTP audio format 0
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 8
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 101
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found audio description format PCMU for ID 0
[Jun 13 00:31:15] Found audio description format PCMA for ID 8
[Jun 13 00:31:15] Found audio description format telephone-event for ID 101
[Jun 13 00:31:15] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 100 Trying
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-PTSJ-1e62261c-21334a3c;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721084 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:15] Audio is at 19088
[Jun 13 00:31:15] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:15] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:15] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-PTSJ-1e62261c-21334a3c;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721084 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] Content-Length: 290
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=root 1445945467 1445945471 IN IP4 82.222.333.444
[Jun 13 00:31:15] s=Asterisk PBX 11.7.0
[Jun 13 00:31:15] c=IN IP4 82.222.333.444
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-16
[Jun 13 00:31:15] a=silenceSupp:off - - - -
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:15] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] ACK sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721084 ACK
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-IDMF-1e622627-282369de
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (9 headers 0 lines) ---
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '083fec101274a55057afb16861e87893@82.222.333.444:15060' of Response 226721084: Match Found
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- SIP read from UDP:91.111.222.33:5060 --->
[Jun 13 00:31:15] INVITE sip:0033123456789@82.222.333.444:15060 SIP/2.0
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Contact: <sip:10.7.1.68:5060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] CSeq: 226721085 INVITE
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] Max-Forwards: 30
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ZAWR-1e62262a-52f3ab17
[Jun 13 00:31:15] Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Jun 13 00:31:15] User-Agent: Cirpack/v4.56 (gw_sip)
[Jun 13 00:31:15] Content-Length: 265
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=cp10 143414827241 143414827249 IN IP4 10.7.1.154
[Jun 13 00:31:15] s=SIP Call
[Jun 13 00:31:15] c=IN IP4 91.121.129.146
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 33122 RTP/AVP 0 8 101
[Jun 13 00:31:15] b=AS:82
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000/1
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000/1
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-15
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15] <------------->
[Jun 13 00:31:15] --- (13 headers 13 lines) ---
[Jun 13 00:31:15] Sending to 91.111.222.33:5060 (NAT)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:25270 handle_request_invite: Initializing initreq for method INVITE - callid 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] Found RTP audio format 0
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 8
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found RTP audio format 101
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:31:15] Found audio description format PCMU for ID 0
[Jun 13 00:31:15] Found audio description format PCMA for ID 8
[Jun 13 00:31:15] Found audio description format telephone-event for ID 101
[Jun 13 00:31:15] Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jun 13 00:31:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] Peer audio RTP is at port 91.121.129.146:33122
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf00035e8'
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:10766 process_sdp: Setting native formats after processing SDP. peer joint formats (ulaw|alaw), old nativeformats (alaw)
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 100 Trying
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ZAWR-1e62262a-52f3ab17;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721085 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Length: 0
[Jun 13 00:31:15]
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:31:15] DEBUG[24926][C-00000003]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:31:15] Audio is at 19088
[Jun 13 00:31:15] Adding codec 100004 (alaw) to SDP
[Jun 13 00:31:15] Adding codec 100003 (ulaw) to SDP
[Jun 13 00:31:15] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 13 00:31:15]
[Jun 13 00:31:15] <--- Reliably Transmitting (NAT) to 91.111.222.33:5060 --->
[Jun 13 00:31:15] SIP/2.0 200 OK
[Jun 13 00:31:15] Via: SIP/2.0/UDP 91.111.222.33:5060;branch=z9hG4bK-ZAWR-1e62262a-52f3ab17;received=91.111.222.33;rport=5060
[Jun 13 00:31:15] Record-Route: <sip:91.111.222.33:5060;lr>
[Jun 13 00:31:15] From: <sip:0123456789@sip.ovh.fr>;tag=00-08187-0f1957b9-566d2b160
[Jun 13 00:31:15] To: "David-HP" <sip:0033123456789@82.222.333.444:15060>;tag=as54293e07
[Jun 13 00:31:15] Call-ID: 083fec101274a55057afb16861e87893@82.222.333.444:15060
[Jun 13 00:31:15] CSeq: 226721085 INVITE
[Jun 13 00:31:15] Server: Asterisk PBX 11.7.0
[Jun 13 00:31:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jun 13 00:31:15] Supported: replaces, timer
[Jun 13 00:31:15] Contact: <sip:0033123456789@82.222.333.444:15060>
[Jun 13 00:31:15] Content-Type: application/sdp
[Jun 13 00:31:15] Content-Length: 290
[Jun 13 00:31:15]
[Jun 13 00:31:15] v=0
[Jun 13 00:31:15] o=root 1445945467 1445945472 IN IP4 82.222.333.444
[Jun 13 00:31:15] s=Asterisk PBX 11.7.0
[Jun 13 00:31:15] c=IN IP4 82.222.333.444
[Jun 13 00:31:15] t=0 0
[Jun 13 00:31:15] m=audio 19088 RTP/AVP 8 0 101
[Jun 13 00:31:15] a=rtpmap:8 PCMA/8000
[Jun 13 00:31:15] a=rtpmap:0 PCMU/8000
[Jun 13 00:31:15] a=rtpmap:101 telephone-event/8000
[Jun 13 00:31:15] a=fmtp:101 0-16
[Jun 13 00:31:15] a=silenceSupp:off - - - -
[Jun 13 00:31:15] a=ptime:20
[Jun 13 00:31:15] a=sendrecv
[Jun 13 00:31:15]
[Jun 13 00:31:15] <------------>
[Jun 13 00:31:15] DEBUG[25190][C-00000003]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0022f50 -- Probation learning mode pass with source address 91.121.129.146:33122
> [Jun 13 00:31:15] > 0x7f6cf0022f50 -- Probation passed - setting RTP source address to 91.121.129.146:33122

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

Asterisk places a call with some music on hold. Callee picks up. music on hold stops as expected then callee hangs up. Then asterisk starts music on hold again for 5 seconds ???
Any idea as to why this would happen . It does not happen if caller hangs up.
Thanks for providing any clue

Called SIP/0XXXXXXXX@forfait-ovh
-- [Jun 12 19:29:48] -- Started music on hold, class 'default', on SIP/davidhp-0000003c
> [Jun 12 19:29:49] > 0x7f621c03a270 -- Probation passed - setting RTP source address to 192.168.0.106:61362
-- [Jun 12 19:29:49] -- SIP/forfait-ovh-0000003d is ringing
-- [Jun 12 19:29:49] -- SIP/forfait-ovh-0000003d is making progress passing it to SIP/davidhp-0000003c
> [Jun 12 19:29:49] > 0x7f6220014370 -- Probation passed - setting RTP source address to xxxxx
-- [Jun 12 19:29:53] -- SIP/forfait-ovh-0000003d answered SIP/davidhp-0000003c
-- [Jun 12 19:29:53] -- Stopped music on hold on SIP/davidhp-0000003c
-- [Jun 12 19:29:53] -- Locally bridging SIP/davidhp-0000003c and SIP/forfait-ovh-0000003d
-- [Jun 12 19:29:53] -- Locally bridging SIP/davidhp-0000003c and SIP/forfait-ovh-0000003d
> [Jun 12 19:29:53] > 0x7f6220014370 -- Probation passed - setting RTP source address to xxx
> [Jun 12 19:29:53] > 0x7f6220014370 -- Probation passed - setting RTP source address to xxx
> [Jun 12 19:29:53] > 0x7f6220014370 -- Probation passed - setting RTP source address to xxx
> [Jun 12 19:29:53] > 0x7f6220014370 -- Probation passed - setting RTP source address to xxx
> [Jun 12 19:29:53] > 0x7f6220014370 -- Probation passed - setting RTP source address to xxx
> [Jun 12 19:29:53] > 0x7f6220014370 -- Probation passed - setting RTP source address to xxx
-- [Jun 12 19:29:55] -- Started music on hold, class 'default', on SIP/davidhp-0000003c

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

You will need to enable SIP packet logging by doing "sip set debug on" and then providing the output to confirm that the signaling is not resulting in hold.

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

In particular, when you hang up the called party on an analogue line, the ISDN network sends CLEAR. Some ISDN to SIP gateways will treat this as hold (as will some mobile networks). If the phone is picked up again (the idea is you pick up a parallel extension in another room) the network will send REANSWER, which will be treated as unhold. Eventually the network will send RELEASE to finally close the call, but this can take several minutes. Some gateways will respond to CLEAR with RELEASE and close the call immediately.

Note that people using Asterisk directly to analogue lines won't see anything for CLEAR.

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

As more clue it indeed does not occur when asterisk calls a sip account
attached are debug messages:
an_sip.c:5736 do_setnat: Setting NAT on RTP to On
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '73f2d5ca00e1fc1a46e29afe46be9ed8@127.0.1.1:15060' to '71edaf7711425f407801d62625715a43@82.222.333.444:15060'
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: dsp.c:482 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: dsp.c:482 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:6507 ast_channel_inherit_variables: Not copying variable SIPURI.
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: chan_sip.c:6356 sip_call: Outgoing Call for 0123456789
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (gsm|ulaw|alaw) Video flag: False Text flag: False
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method INVITE - callid 71edaf7711425f407801d62625715a43@82.222.333.444:15060
-- [Jun 13 00:18:59] -- Called SIP/0123456789@forfait-ovh
-- [Jun 13 00:18:59] -- Started music on hold, class 'default', on SIP/davidhp-00000004
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:3577 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:3740 ast_read_generator_actions: Generator got voice, switching to phase locked mode
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:3577 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: channel.c:5361 set_format: Set channel SIP/davidhp-00000004 to write format slin
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: res_musiconhold.c:355 ast_moh_files_next: SIP/davidhp-00000004 Opened file 0 '/var/lib/asterisk/moh/transfert.2014'
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2707 ast_rtp_write: Ooh, format changed from unknown to alaw
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2742 ast_rtp_write: Created smoother: format: alaw ms: 20 len: 160
[Jun 13 00:18:59] DEBUG[24926][C-00000002]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' Request 102: Found
[Jun 13 00:18:59] DEBUG[24926][C-00000002]: chan_sip.c:4529 __sip_ack: Acked pending invite 102
[Jun 13 00:18:59] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Request 102: Match Found
[Jun 13 00:18:59] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (gsm|ulaw|alaw) Video flag: False Text flag: False
[Jun 13 00:18:59] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:18:59] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 80 bytes
[Jun 13 00:18:59] DEBUG[24926][C-00000002]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' Request 103: Found
[Jun 13 00:19:00] DEBUG[24926][C-00000002]: chan_sip.c:4608 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' Request 103: Found
[Jun 13 00:19:00] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dc910
[Jun 13 00:19:00] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dc910
[Jun 13 00:19:00] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dc910
[Jun 13 00:19:00] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:00] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
-- [Jun 13 00:19:00] -- SIP/forfait-ovh-00000005 is ringing
-- [Jun 13 00:19:00] -- SIP/forfait-ovh-00000005 is making progress passing it to SIP/davidhp-00000004
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 80 bytes
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:4529 __sip_ack: Acked pending invite 103
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Request 103: Match Found
-- [Jun 13 00:19:03] -- SIP/forfait-ovh-00000005 answered SIP/davidhp-00000004
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: rtp_engine.c:1805 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/davidhp-00000004' with that of 'SIP/forfait-ovh-00000005'
-- [Jun 13 00:19:03] -- Stopped music on hold on SIP/davidhp-00000004
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: channel.c:5361 set_format: Set channel SIP/davidhp-00000004 to write format alaw
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: channel.c:3577 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: features.c:4250 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/forfait-ovh-00000005 since we're bridging
-- [Jun 13 00:19:03] -- Locally bridging SIP/davidhp-00000004 and SIP/forfait-ovh-00000005
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706434: Match Found
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706435: Match Found
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706436: Match Found
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0003590 -- Probation learning mode pass with source address 91.121.129.146:37072
> [Jun 13 00:19:03] > 0x7f6cf0003590 -- Probation passed - setting RTP source address to 91.121.129.146:37072
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0003590 -- Probation learning mode pass with source address 91.121.129.146:37072
> [Jun 13 00:19:03] > 0x7f6cf0003590 -- Probation passed - setting RTP source address to 91.121.129.146:37072
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706437: Match Found
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0003590 -- Probation learning mode pass with source address 91.121.129.146:37072
> [Jun 13 00:19:03] > 0x7f6cf0003590 -- Probation passed - setting RTP source address to 91.121.129.146:37072
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706438: Match Found
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:03] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:03] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0003590 -- Probation learning mode pass with source address 91.121.129.146:37072
> [Jun 13 00:19:03] > 0x7f6cf0003590 -- Probation passed - setting RTP source address to 91.121.129.146:37072
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706439: Match Found
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0003590 -- Probation learning mode pass with source address 91.121.129.146:37072
> [Jun 13 00:19:04] > 0x7f6cf0003590 -- Probation passed - setting RTP source address to 91.121.129.146:37072
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706440: Match Found
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0003590 -- Probation learning mode pass with source address 91.121.129.146:37072
> [Jun 13 00:19:04] > 0x7f6cf0003590 -- Probation passed - setting RTP source address to 91.121.129.146:37072
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706441: Match Found
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706442: Match Found
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3581 ast_rtp_read: 0x7f6cf0003590 -- Probation learning mode pass with source address 91.121.129.146:37072
> [Jun 13 00:19:04] > 0x7f6cf0003590 -- Probation passed - setting RTP source address to 91.121.129.146:37072
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7f6cc30dd4d0
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3913 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13596 transmit_response_with_sdp: Setting framing from config on incoming call
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13151 add_sdp: ** Our capability: (alaw) Video flag: True Text flag: True
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:13152 add_sdp: ** Our prefcodec: (alaw)
-- [Jun 13 00:19:04] -- Started music on hold, class 'default', on SIP/davidhp-00000004
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: channel.c:3577 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: channel.c:3740 ast_read_generator_actions: Generator got voice, switching to phase locked mode
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: channel.c:3577 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: channel.c:5361 set_format: Set channel SIP/davidhp-00000004 to write format slin
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_musiconhold.c:355 ast_moh_files_next: SIP/davidhp-00000004 Opened file 0 '/var/lib/asterisk/moh/transfert.2014'
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '71edaf7711425f407801d62625715a43@82.222.333.444:15060' of Response 226706443: Match Found
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:04] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
......
[Jun 13 00:19:07] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:07] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:07] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3184 ast_rtcp_read: Got RTCP report of 80 bytes
[Jun 13 00:19:07] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:07] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:07] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:08] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:2675 ast_rtp_write: No remote address on RTP instance '0x7f6cf0021da8' so dropping frame
[Jun 13 00:19:09] DEBUG[24926][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: rtp_engine.c:1033 local_bridge_loop: rtp-engine-local-bridge: Ooh, got a hangup
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: channel.c:7992 ast_channel_bridge: Returning from native bridge, channels: SIP/davidhp-00000004, SIP/forfait-ovh-00000005
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: channel.c:2840 ast_hangup: Hanging up channel 'SIP/forfait-ovh-00000005'
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: chan_sip.c:7060 sip_hangup: Hangup call SIP/forfait-ovh-00000005, SIP callid 71edaf7711425f407801d62625715a43@82.222.333.444:15060
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: app_dial.c:3100 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: pbx.c:6573 __ast_pbx_run: Spawn extension (test,9,3) exited non-zero on 'SIP/davidhp-00000004'
== [Jun 13 00:19:09] == Spawn extension (test, 9, 3) exited non-zero on 'SIP/davidhp-00000004'
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: channel.c:2661 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/davidhp-00000004'
-- [Jun 13 00:19:09] -- Stopped music on hold on SIP/davidhp-00000004
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: channel.c:5361 set_format: Set channel SIP/davidhp-00000004 to write format alaw
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: channel.c:2840 ast_hangup: Hanging up channel 'SIP/davidhp-00000004'
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: chan_sip.c:7060 sip_hangup: Hangup call SIP/davidhp-00000004, SIP callid 76589OWUxMmEyMGUxMWIzOWM2Njc4ZGNjMTdhODNiOThlNDE
[Jun 13 00:19:09] DEBUG[24948]: app_queue.c:1807 handle_statechange: Device 'SIP/davidhp' changed to state '1' (Not in use)
[Jun 13 00:19:09] DEBUG[25090][C-00000002]: res_rtp_asterisk.c:3994 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f6ce0002df8'
[Jun 13 00:19:09] DEBUG[24948]: app_queue.c:1807 handle_statechange: Device 'SIP/davidhp' changed to state '1' (Not in use)
[Jun 13 00:19:09] DEBUG[24926][C-00000002]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '76589OWUxMmEyMGUxMWIzOWM2Njc4ZGNjMTdhODNiOThlNDE' of Request 102: Match Found
[Jun 13 00:19:09] DEBUG[24926]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7f6ce0002df8'
[Jun 13 00:19:16] DEBUG[24926]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x7f6cf0021da8'
[Jun 13 00:19:18] DEBUG[24926]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 4df6b9d317a45ce903bb76c32f9a2a14@127.0.1.1:15060 - OPTIONS (No RTP)
[Jun 13 00:19:18] DEBUG[24926]: chan_sip.c:3999 ast_sip_ouraddrfor: Target address 91.111.222.33:5060 is not local, substituting externaddr
[Jun 13 00:19:18] DEBUG[24926]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '4df6b9d317a45ce903bb76c32f9a2a14@127.0.1.1:15060' to '54b413e37a3ea81a31620c293f6586e9@82.222.333.444:15060'
[Jun 13 00:19:18] DEBUG[24926]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 54b413e37a3ea81a31620c293f6586e9@82.222.333.444:15060
[Jun 13 00:19:18] DEBUG[24926]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '54b413e37a3ea81a31620c293f6586e9@82.222.333.444:15060' of Request 102: Match Found
[Jun 13 00:19:27] DEBUG[24926]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 5b6c04c90e39e2406acb60e64d0903be@127.0.1.1 - REGISTER (No RTP)
[Jun 13 00:19:27] DEBUG[24926]: chan_sip.c:3999 ast_sip_ouraddrfor: Target address 91.111.222.33:5060 is not local, substituting externaddr
[Jun 13 00:19:27] DEBUG[24926]: chan_sip.c:15296 transmit_register: Scheduled a registration timeout for sip.ovh.fr id #174
[Jun 13 00:19:27] DEBUG[24926]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method REGISTER - callid 5b6c04c90e39e2406acb60e64d0903be@127.0.1.1
[Jun 13 00:19:28] DEBUG[24926]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '5b6c04c90e39e2406acb60e64d0903be@127.0.1.1' of Request 111: Match Found
[Jun 13 00:19:28] DEBUG[24926]: chan_sip.c:3516 initialize_initreq: Initializing already initialized SIP dialog 5b6c04c90e39e2406acb60e64d0903be@127.0.1.1 (presumably reinvite)
[Jun 13 00:19:28] DEBUG[24926]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '5b6c04c90e39e2406acb60e64d0903be@127.0.1.1' of Request 112: Match Found
[Jun 13 00:19:28] DEBUG[24926]: chan_sip.c:23419 handle_response_register: Registration successful
[Jun 13 00:19:28] DEBUG[24926]: chan_sip.c:23421 handle_response_register: Cancelling timeout 174
[Jun 13 00:20:18] DEBUG[24926]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 47bdab6d48ff54a17bf2a7c9657e8f07@127.0.1.1:15060 - OPTIONS (No RTP)
[Jun 13 00:20:18] DEBUG[24926]: chan_sip.c:3999 ast_sip_ouraddrfor: Target address 91.111.222.33:5060 is not local, substituting externaddr
[Jun 13 00:20:18] DEBUG[24926]: chan_sip.c:8565 change_callid_pvt: SIP call-id changed from '47bdab6d48ff54a17bf2a7c9657e8f07@127.0.1.1:15060' to '54c0c5d1487a9b2009062c710035b2c0@82.222.333.444:15060'
[Jun 13 00:20:18] DEBUG[24926]: chan_sip.c:3518 initialize_initreq: Initializing initreq for method OPTIONS - callid 54c0c5d1487a9b2009062c710035b2c0@82.222.333.444:15060
[Jun 13 00:20:18] DEBUG[24926]: chan_sip.c:4567 __sip_ack: Stopping retransmission on '54c0c5d1487a9b2009062c710035b2c0@82.222.333.444:15060' of Request 102: Match Found
[Jun 13 00:20:28] DEBUG[24926]: chan_sip.c:3999 ast_sip_ouraddrfor: Target address 91.111.222.33:5060 is not local, substituting externaddr
[Jun 13 00:20:28] DEBUG[24926]: chan_sip.c:8770 sip_alloc: Allocating new SIP dialog for 00-03398-3539bacf-778af0826@91.111.222.33 - OPTIONS (No RTP)

Statistics : Posted by drtpro • on Fri Jun 12, 2015 11:33 am • Replies 6 • Views 174

---

Hello everybody.

I have what I think could be a very basic configuration problem.
I am trying to setup an 8 port analog gateway. 4 ports are FXS and the other are FXO.
My idea is to register each port individually so for example the 4 FXS can belong to an internal context, and the 4 FXO can belong to an external one (may be "FROM-PSTN").
If I do that, almost everything works fine... I can call my bria softphone from any of the FXS and vice versa. I can also call from the softphone to the PSTN via the FXOs.
The problem I have is related to the incoming calls from the PSTN. Asterisk sees the incoming calls as coming from unknown endpoints, and this makes sense to me because every caller has a different caller ID...
This is what I am talking about:
res_pjsip/pjsip_distributor.c:256 log_unidentified_request: Request from '"11XXXXX539" <sip:11XXXXX539@192.168.101.161>' failed for '192.168.101.161:64387' (callid: 184590733321201023112@192.168.101.161) - No matching endpoint found

When in the past I had and analog gateway with only one FXO, I solved this using the IDENTIFY parameter with a match to the IP of my old gateway.
Now I can't do that because each port is registering from the same IP address, so the identify wouldn't make any sense here.

Can anyone tell me how should I setup my gateway or what is the recommended way to setup this kind of gateways with Asterisk?


The other option that I am analyzing consists into registering the gateway as a whole. This will work for incoming and outgoing calls, but this way the 8 ports will be in the same context and I don't like that.

Any help will be appreciated.

Thanks

Gaston.

Statistics : Posted by gtheaded • on Fri Jun 12, 2015 9:29 pm • Replies 0 • Views 59

---

I have been playing around with the following sample genericMenu example. I want to call a genericForm when the select button is pressed. However nothing happens. The menu keep functioning in a never ending loop.

Code:

var genericMenu = require('genericMenu');
var handler = {};
//This handler is called when any assigned softkeys/hardkeys are pressed
//It receives the id of the selected item and the action as parameters
handler.processMenuAction = function (params) {
   switch (params.actionId) {
   case 'select':
      selectCallback(params.selectionId);
      break;
   case 'exit':
      digium.background();
      break;
   }
};
//the menu items are defined in an array
var items = [
   {
      'text'  : 'Menu Entry 1',
      'id'    : 'item_1'
   },
   {
      'text'  : 'Menu Entry 2',   
      'id'    : 'item_2'
   },
   {
      'text'  : 'Menu Entry 3',
      'id'    : 'item_3'
   },
   {
      'text'  : 'Menu Entry 4',
      'id'    : 'item_4'
   }
];
//the softkeys are defined in an array
var softkeys = [
   {
      'label'      : 'Select',
      'actionId'   : 'select',
      'icon'      : app.images.softKeys.select
   },
   {
      'label'      : 'Cancel',
      'actionId'   : 'exit'
   }
];
//show the menu
genericMenu.show({
   'id'      : 'menu_1',
   'menu'      : items,
   'object'   : handler,
   'title'      : 'Menu Title',
   'softkeys'   : softkeys,
   'onkeyselect'   : selectCallback,
   'onkeycancel'   : digium.background
        'forceRedraw'   : true
});

function selectCallback(selection) {
    util.debug("Selected " + selection);
   };

I have tried calling my function for the form within the callback function as follows.

Code:

function selectCallback(selection) {
    util.debug("Selected " + selection);
    myForm();
   };

I have also tried adding a statement to stop observing the event as follows but all to no avail

Code:

function selectCallback(selection) {
    util.debug("Selected " + selection);
   digium.event.stopObserving({
   'eventName' : 'digium.app.foreground'
   });
   myForm();
   };

I have confirmed that the code within the form function is being executed. However teh menu remains on the screen and the keypress event keeps firing.

Statistics : Posted by intellectit • on Sun Jun 14, 2015 4:49 am • Replies 0 • Views 35

---

I just installed a new system here is my setup

Asterisk 11
Digium T132
Dahdi version is 2.11.0.1
FreePBX version 6.12.65.28

When a call comes in initially there is a 3 to 4 second lag time in the audio. This is only for the first talker in the call. If the call is transferred to another phone the call is perfect. Initially we had some issues with getting the PRI to connect to the provider. The provider ssh'd into my system and found that a lib file was conflicting with my version of Asterisk. He changed it and everything started to work. This installation is not even a week old. I know there is a newer DAHDI driver available. The question is: Is there a known issue with the driver I am using with Asterisk 11 and if so how do I go about updating to the new one.


Thanks for any help.

Statistics : Posted by edlentz • on Sun Jun 14, 2015 7:33 am • Replies 0 • Views 28

---

The purpose of endpoint identification is to determine who a SIP message is coming from. Right now there's 4 ways:

1. The user portion of the 'From' header
2. By source IP address
3. By a 'line' parameter which can be enabled in an outbound registration
4. Anonymously

If you can't differentiate on a per-port basis using the above then the only fallback would be matching based on IP address, sending it to a context, and then using PJSIP_HEADER to extract specific information that identifies the port. Based on that you could Goto() elsewhere.

That requires there to be some sort of unique identifying information in the SIP message. Is there?

If not then I'm afraid you are out of luck.

Statistics : Posted by gtheaded • on Fri Jun 12, 2015 9:29 pm • Replies 1 • Views 138

---