Hello everyone

How do I set the Asterisk to close any prompt sounds(.gsm, .wav....) or warming tone?

thanks!!

Statistics : Posted by adolclistin • on Thu Mar 28, 2013 1:17 am • Replies 0 • Views 4

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You need to compile asterisk with that channel driver. You need to perform the make menuselect and then enable the chan_console.

Statistics : Posted by Lexus45 • on Mon Mar 25, 2013 5:41 am • Replies 5 • Views 100

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While everything worked well one one machine, I've faced the problem on the other one.

Code:

WARNING[3422]: channel.c:5627 ast_request: No channel type registered for 'CONSOLE'
WARNING[3422]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'CONSOLE' (cause 66 - Channel not implemented)

The sound card works, because this machine 'sings songs' in the WC
Config files are similar. Modules loaded are similar.
This is a RaspberryPi. Maybe that is why it doesn't work ?

Statistics : Posted by Lexus45 • on Mon Mar 25, 2013 5:41 am • Replies 5 • Views 100

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This works

; 1002
exten => 1002,1,Dial(CONSOLE/ALSA)
exten => 1002,n,Hangup()


After dialing [1002] from an IP-phone, I can hear/talk to the dialer from the microphone and speakers, connected to the Asterisk box directly.
It's interesting that I haven't edited alsa.conf at all.

Statistics : Posted by Lexus45 • on Mon Mar 25, 2013 5:41 am • Replies 5 • Views 100

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I think I should start from here:
http://www.voip-info.org/wiki/view/Aste ... +alsa.conf
or
http://www.voip-info.org/wiki/view/Aste ... g+oss.conf

Statistics : Posted by Lexus45 • on Mon Mar 25, 2013 5:41 am • Replies 5 • Views 100

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Hello all.
I'm not sure if I formulated the question well enough. I hope you will understand.
What is the best way to 'redirect' the sound of a phone call to the sound card?

The situation is as follows - I need to set up a server which will notify loudly some group of people in the room, where it will be located.

So, after some extension is dialed, this Asterisk/softphone answers automatically, and everything that is said to the dialed extension, will be heard by the audience.

I'm not sure if Asterisk is the right solution, maybe some softphone will be OK (we have the 'main' Asterisk server where extensions are stored in the config file).

Maybe all I need - is to know how to forward all output sound to the sound card. If I'm not mistaken , chan_oss is able to do something like this. Is there any easier and more simple way to achieve the result ?

Say, that extension is [1111]. When somebody dials [1111], that PC with Asterisk/softphone answers automatically and everybody hear what is said to that channel.

Share your ideas please. Any of them will be useful!

Statistics : Posted by Lexus45 • on Mon Mar 25, 2013 5:41 am • Replies 5 • Views 100

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I use Asterisk 10.7.0 and "nat=yes" (Force rport to always be on and perform comedia RTP handling).

asterisk -rx "sip show settings"

Code:

Parsing /etc/asterisk/extconfig.conf


Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        No
  Match Auth Username:    Yes
  Allow unknown access:   No
  Allow subscriptions:    No
  Allow overlap dialing:  No
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             Asterisk 10.7.0
  SDP Session Name:       Asterisk 10.7.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T.38 support:           Yes
  T.38 EC mode:           Redundancy
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10

Global Signalling Settings:
---------------------------
  Codecs:                 (gsm|ulaw|alaw|h263|testlaw)
  Codec Order:            none
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            60
  RTP Hold Timeout:       120
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  SIP Transfer mode:      closed
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                default
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               ru
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk

----
Asterisk ending (0).

asterisk -rx "sip show peer r1234_line1"

Code:

Parsing /etc/asterisk/extconfig.conf


  * Name       : r1234_line1
  Description  :
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : from-ipclients
  Subscr.Cont. : <Not set>
  Language     : ru
  AMA flags    : Unknown
  Transfer mode: closed
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  MOH Suggest  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 1
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "" <1234567890>
  MaxCallBR    : 384 kbps
  Expire       : 3218
  Insecure     : no
  Force rport  : Yes
  ACL          : No
  DirectMedACL : No
  T.38 support : Yes
  T.38 EC mode : Redundancy
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: No
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 95.138.160.45:62893
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: r1234_line1
  SIP Options  :
  Codecs       : (gsm|ulaw|alaw|g729)
  Codec Order  : (alaw:20,ulaw:20,g729:20,gsm:20)
  Auto-Framing :  No
  Status       : OK (160 ms)
  Useragent    : Linksys/SPA2102-3.3.6
  Reg. Contact : sip:r1234_line1@192.168.5.17:5060
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

Asterisk ending (0)

tcpdump

Code:

09:28:20.932560 IP 95.138.160.45.62893 > 88.80.1.50.5060: SIP, length: 422
09:28:24.681426 IP 88.80.1.50.5060 > 95.138.160.45.62893: SIP, length: 945
09:28:24.924838 IP 95.138.160.45.62893 > 88.80.1.50.5060: SIP, length: 322
09:28:24.925086 IP 95.138.160.45.62893 > 88.80.1.50.5060: SIP, length: 433
09:28:27.264832 IP 95.138.160.45.62893 > 88.80.1.50.5060: SIP, length: 814
09:28:27.268305 IP 88.80.1.50.5060 > 192.168.5.17.5060: SIP, length: 442
09:28:27.389518 IP 88.80.1.50.15200 > 192.168.5.17.16400: UDP, length 32
09:28:27.410068 IP 88.80.1.50.15200 > 192.168.5.17.16400: UDP, length 32

Why media traffic going to 192.168.5.17 ?

Statistics : Posted by axonaro • on Thu Mar 28, 2013 11:27 pm • Replies 0 • Views 7

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Well they should be on Freepbx forum... but i installed Asterisk Now... and FreePbx is in it... is it not the right forum ??

Statistics : Posted by hassancheema36 • on Thu Mar 28, 2013 1:33 am • Replies 2 • Views 36

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Questions about configuring FreePBX should be directed to http://www.freepbx.org/forums

Statistics : Posted by hassancheema36 • on Thu Mar 28, 2013 1:33 am • Replies 2 • Views 36

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Any one Please help if there is any way to Set IVR or Auto Attendant ... Please Please Help

Statistics : Posted by hassancheema36 • on Thu Mar 28, 2013 1:33 am • Replies 2 • Views 36

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Hi, All!
We use Digium Switchvox SMB 305
For test I use softphone X-lite ver 4.5.
We use SIP. We DO NOT use ISDN PRI
I have problem with outgoing call creating
I'm trying to create an outgoing call, I hear something like: "All socket are busy. Error id 5".
In "Call logs" I see

Code:

--------------------------------------------------------
Event Name     Time          Details
--------------------------------------------------------
Outgoing       12:17:41 AM   Dialed number (9XXXXXXXXX)
Provider       12:17:41 AM   Sent call over SIP Provider ( VoicePulse ) with number XXXXXXXXX
Status         12:17:42 AM   Received status of CONGESTION with a cause code of 34
Hang up        12:17:47 AM   Call was hung up by the PBX ( Unknown Account )
--------------------------------------------------------

In "Error Logs" I see:
SIP call failed when dialing 9XXXXXXXXX using VoicePulse. Check the status of the provider on the System Status page.

In what could be the problem?

Statistics : Posted by sir_genry • on Fri Mar 29, 2013 12:38 am • Replies 0 • Views 7

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LOOKS LIKE NO ONE WANTS TO TAKE THE TIME TO TRAVERSE THROUGH MASSIVE LOG FILES. OH WELL. ALSO, YOU'RE ON AN OLD VERSION.

Statistics : Posted by shabbir92 • on Fri Dec 07, 2012 1:43 pm • Replies 5 • Views 348

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can anyone help me to solve this issue ????????

Statistics : Posted by shabbir92 • on Fri Dec 07, 2012 1:43 pm • Replies 5 • Views 348

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This is detailed log of a failed call


[Dec 8 12:03:33] DEBUG[3523]: logger.c:1285 ast_create_callid: CALL_ID [C-00000000] created by thread.
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '0.0.0.0' into...
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '0.0.0.0' and port ''.
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xb732b164'
[Dec 8 12:03:33] DEBUG[3523]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 19592 for RTP instance '0xb732b164'
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.208' into...
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.208' and port ''.
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0xb732b164' is setup and ready to go
[Dec 8 12:03:33] DEBUG[3523]: res_rtp_asterisk.c:3848 ast_rtp_prop_set: Setup RTCP on RTP instance '0xb732b164'
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.208' into...
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.208' and port ''.
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for Motif -xxxxxx
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for Motif/xxxxxx - state 2 (In use)
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:440 devstate_event: device 'Motif/xxxxxx' state '2'
[Dec 8 12:03:33] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device 'Motif/xxxxxx' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 102 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 100 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 106 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 100 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 102 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 106 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '195.211.49.73' into...
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '195.211.49.73' and port ''.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: pbx.c:4410 pbx_extension_helper: Launching 'NoOp'
-- Executing [s@incoming-11111:1] NoOp("Motif/xxxxxx-3a32", "") in new stack
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: pbx.c:4410 pbx_extension_helper: Launching 'Dial'
-- Executing [s@incoming-11111:2] Dial("Motif/xxxxxx-3a32", "SIP/200,90,D(:1)") in new stack
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:29048 sip_request_call: Asked to create a SIP channel with formats: (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:8401 sip_alloc: Allocating new SIP dialog for 20fe81b17dabebbf26ac47f52f39c768@192.168.15.208:5060 - INVITE (No RTP)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xb750c73c'
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 14872 for RTP instance '0xb750c73c'
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.208' into...
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.208' and port ''.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0xb750c73c' is setup and ready to go
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:33] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:3848 ast_rtp_prop_set: Setup RTCP on RTP instance '0xb750c73c'
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.208' into...
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.208' and port ''.
== Using SIP RTP CoS mark 5
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:5420 do_setnat: Setting NAT on RTP to Off
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: acl.c:979 ast_ouraddrfor: For destination '192.168.15.200', our source address is '192.168.15.208'.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:3744 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.15.208:5060
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: format_pref.c:339 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7589 sip_new: *** Our native formats are (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7590 sip_new: *** Joint capabilities are (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7591 sip_new: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7592 sip_new: *** AST_CODEC_CHOOSE formats are ulaw
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7594 sip_new: *** Our preferred formats from the incoming channel are (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7620 sip_new: This channel will not be able to handle video.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel_internal_api.c:860 ast_channel_callid_set: Channel Call ID changing from [C-00000000] to [C-00000000]
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:6028 sip_call: Outgoing Call for 200
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:6352 update_call_counter: Updating call counter for outgoing call
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12670 add_sdp: This call needs video offers, but there's no video support enabled!
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12718 add_sdp: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12719 add_sdp: ** Our prefcodec: (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12856 add_sdp: -- Done with adding codecs to SDP
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:13059 add_sdp: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|h263|testlaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:3230 initialize_initreq: Initializing initreq for method INVITE - callid 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:3587 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.15.200:5070
-- Called SIP/200
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:3893 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xb732b164'
[Dec 8 12:03:33] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:4320 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060' Request 102: Found
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:22002 handle_response_invite: SIP response 180 to standard invite
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
-- SIP/200-00000000 is ringing
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 200
[Dec 8 12:03:33] DEBUG[3517]: chan_sip.c:28948 sip_devicestate: Checking device state for peer 200
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for SIP/200 - state 1 (Not in use)
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:440 devstate_event: device 'SIP/200' state '1'
[Dec 8 12:03:33] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Dec 8 12:03:33] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:4320 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060' Request 102: Found
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:22002 handle_response_invite: SIP response 180 to standard invite
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
-- SIP/200-00000000 is ringing
[Dec 8 12:03:33] DEBUG[3524]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3524]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:34] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:4241 __sip_ack: Acked pending invite 102
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:4279 __sip_ack: Stopping retransmission on '5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060' of Request 102: Match Found
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:22002 handle_response_invite: SIP response 200 to standard invite
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP o=NCHSoftware-Talk 1354947830 1354947832 IN IP4 192.168.15.200... UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP s=Express Talk Call... UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.200' into...
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.200' and port ''.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP c=IN IP4 192.168.15.200... OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: res_rtp_asterisk.c:3893 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xb750c73c'
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 3 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: res_rtp_asterisk.c:3814 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0xb750c73c'
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10347 process_sdp: We're settling with these formats: (gsm|ulaw|alaw)
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10354 process_sdp: We have an owner, now see if we need to change this call
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: format_pref.c:339 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:6352 update_call_counter: Updating call counter for outgoing call
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:15774 build_route: build_route: Contact hop: <sip:200@192.168.15.200:5070>
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.200:5070' into...
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.200' and port '5070'.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.200:5070' into...
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.200' and port '5070'.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:3587 __sip_xmit: Trying to put 'ACK sip:200' onto UDP socket destined for 192.168.15.200:5070
-- SIP/200-00000000 answered Motif/xxxxxx-3a32
-- Sending DTMF '1' to the calling party.
[Dec 8 12:03:34] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 200
[Dec 8 12:03:34] DEBUG[3517]: chan_sip.c:28948 sip_devicestate: Checking device state for peer 200
[Dec 8 12:03:34] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for SIP/200 - state 1 (Not in use)
[Dec 8 12:03:34] DEBUG[3517]: devicestate.c:440 devstate_event: device 'SIP/200' state '1'
[Dec 8 12:03:34] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 -- start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 -- probation = 4, seq = 6844
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3508 ast_rtp_read: 0xb7511690 -- Condition for learning hasn't exited, so reject the frame.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 -- start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 -- probation = 3, seq = 6845
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3508 ast_rtp_read: 0xb7511690 -- Condition for learning hasn't exited, so reject the frame.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 -- start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 -- probation = 2, seq = 6846
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3508 ast_rtp_read: 0xb7511690 -- Condition for learning hasn't exited, so reject the frame.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 -- start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 -- probation = 1, seq = 6847
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3512 ast_rtp_read: 0xb7511690 -- Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for Motif - xxxxxx
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: features.c:4387 ast_bridge_call: bridge answer set, chan answer set
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: features.c:4229 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/200-00000000 since we're bridging
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update
-- Locally bridging Motif/xxxxxx-3a32 and SIP/200-00000000
[Dec 8 12:03:35] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for Motif/xxxxxx - state 2 (In use)
[Dec 8 12:03:35] DEBUG[3517]: devicestate.c:440 devstate_event: device 'Motif/xxxxxx' state '2'
[Dec 8 12:03:35] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device 'Motif/xxxxxx' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Dec 8 12:03:35] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:36] DEBUG[3523][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:36] DEBUG[3523][C-00000000]: chan_motif.c:2374 jingle_action_session_terminate: Hanging up channel 'Motif/xxxxxx-3a32' due to session terminate message with cause '16'
[Dec 8 12:03:36] DEBUG[3523][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:36] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: rtp_engine.c:1029 local_bridge_loop: rtp-engine-local-bridge: Ooh, got a hangup
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:7917 ast_channel_bridge: Returning from native bridge, channels: Motif/xxxxxx-3a32, SIP/200-00000000
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:2830 ast_hangup: Hanging up channel 'SIP/200-00000000'
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: chan_sip.c:6732 sip_hangup: Hangup call SIP/200-00000000, SIP callid 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:3893 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xb750c73c'
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.15.200:5070' into...
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.15.200' and port '5070'.
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: chan_sip.c:3587 __sip_xmit: Trying to put 'BYE sip:200' onto UDP socket destined for 192.168.15.200:5070
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 200
[Dec 8 12:03:36] DEBUG[3517]: chan_sip.c:28948 sip_devicestate: Checking device state for peer 200
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for SIP/200 - state 1 (Not in use)
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:440 devstate_event: device 'SIP/200' state '1'
[Dec 8 12:03:36] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: app_dial.c:3096 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: pbx.c:6090 __ast_pbx_run: Spawn extension (incoming-11111,s,2) exited non-zero on 'Motif/xxxxxx-3a32'
== Spawn extension (incoming-11111, s, 2) exited non-zero on 'Motif/xxxxxx-3a32'
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:2651 ast_softhangup_nolock: Soft-Hanging up channel 'Motif/xxxxxx-3a32'
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:2830 ast_hangup: Hanging up channel 'Motif/xxxxxx-3a32'
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0xb732b164'
[Dec 8 12:03:36] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:36] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:36] DEBUG[3528][C-00000000]: chan_sip.c:4279 __sip_ack: Stopping retransmission on '5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060' of Request 103: Match Found
[Dec 8 12:03:36] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:36] DEBUG[3528]: chan_sip.c:6500 sip_destroy: Destroying SIP dialog 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060
[Dec 8 12:03:36] DEBUG[3528]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0xb750c73c'
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for Motif - xxxxxx
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for Motif/xxxxxx - state 0 (Unknown)
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:440 devstate_event: device 'Motif/xxxxxx' state '0'
[Dec 8 12:03:36] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device 'Motif/xxxxxx' changed to state '0' (Unknown) but we don't care because they're not a member of any queue.

Statistics : Posted by shabbir92 • on Fri Dec 07, 2012 1:43 pm • Replies 5 • Views 348

---

hello dear friends
i am facing a problum and want solution any ideas wellcome
i have installed centos 6.3
asterisk 11.0.1
and i am able to place call and recieve to my friends when my friends are comes online via {pc,s no metter laptop or desktop} gmail and nimbuzz clients calls in and out working fine
but when my friends comes online using thier mobile nimbuzz clients they call me and my phone also rings but when i answer incoming calls call goes ended,and i want to recieve and place calls to my friends on their nimbuzz clients, please help me
here are my configs
xmpp.conf
[11111]
type=client
serverhost=talk.google.com;
;pubsub_node=pubsub.astjab.org;
username=acount1@gmail.com; U
secret=xxxxxxx;
priority=1; R
port=5222; Po
usetls=yes; Use tl
usesasl=yes; Use
;buddy=mogorman@

sip.conf
[200]
type=friend
host=dynamic
username=200
secret=xxxxx
context=demo
canreinvite=no

motif.conf
[xxxxxx]
disallow=all;
allow=ulaw;
context=incoming- 11111;
connection=xxxxxx

extensions.conf
[incoming-11111]
exten => s,1,NoOp()
exten => s,n,Dial(SIP/200,90,D(:1))
exten => s,n(end),Hangup()
include => demo;

and these are logs
dropped incoming call from nimbuzz mobile client
Executing [s@incoming-11111:1] NoOp("Motif/xxxxxx-30e9", "") in new stack
-- Executing [s@incoming-11111:2] Dial("Motif/xxxxxx-30e9", "SIP/200,90,D(:1)") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- SIP/200-00000000 is ringing
-- SIP/200-00000000 answered Motif/xxxxxx-30e9
-- Sending DTMF '1' to the calling party.
-- Locally bridging Motif/xxxxxx-30e9 and SIP/200-00000000
== Spawn extension (incoming-11111, s, 2) exited non-zero on 'Motif/xxxxxx-30e9'

and this is successfull call nimbuzz desktop client

Executing [s@incoming-11111:1] NoOp("Motif/xxxxxx-10ba", "") in new stack
-- Executing [s@incoming-11111:2] Dial("Motif/xxxxxx-10ba", "SIP/200,90,D(:1)") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- SIP/200-00000001 is ringing
-- SIP/200-00000001 is ringing
-- SIP/200-00000001 is ringing
-- SIP/200-00000001 answered Motif/xxxxxx-10ba
-- Sending DTMF '1' to the calling party.
-- Locally bridging Motif/xxxxxx-10ba and SIP/200-00000001
== Spawn extension (incoming-11111, s, 2) exited non-zero on 'Motif/xxxxxx-10ba'
i have edited some email addreses in config and log files
any ideas welcome

Statistics : Posted by shabbir92 • on Fri Dec 07, 2012 1:43 pm • Replies 5 • Views 348

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An additional thing to note, my softphone is on a different subnet and VLAN although I can route to any VLAN, I dont know if this would cause an issue. I have tried this using physical handsets on the same VLAN and got the same result last week.

Statistics : Posted by duxy786 • on Thu Mar 28, 2013 5:26 pm • Replies 5 • Views 72

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Hi David,

Thank you for your quick response. The second call is placed before line 3781. The lines just before and between 3782 to 3803 is when I believe the issue happens resulting in a response from Asterisk with a "no service" Playback. See below:

Code:

<------------->
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  0 [ 14]: SIP/2.0 200 OK
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  1 [ 66]: Via: SIP/2.0/UDP 10.10.0.15:5060;branch=z9hG4bK5fa892d1;rport=5060
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  2 [ 30]: Contact: <sip:10.10.3.64:5060>
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  3 [ 70]: To: <sip:8480@10.10.3.64:5060;rinstance=3ab7f5168605473e>;tag=695d96db
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  4 [ 54]: From: "Unknown"<sip:Unknown@10.10.0.15>;tag=as5062efe9
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  5 [ 57]: Call-ID: 261090773aaf677807b853f65df7c2cd@10.10.0.15:5060
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  6 [ 17]: CSeq: 102 OPTIONS
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  7 [ 23]: Accept: application/sdp
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  8 [ 19]: Accept-Language: en
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header  9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header 10 [ 19]: Supported: replaces
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header 11 [ 44]: User-Agent: X-Lite release 5.0.0 stamp 67284
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c:  Header 12 [ 17]: Content-Length: 0
[Mar 29 20:35:28] VERBOSE[30640] chan_sip.c: --- (13 headers 0 lines) ---
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c: = Looking for  Call ID: 261090773aaf677807b853f65df7c2cd@10.10.0.15:5060 (Checking To) --From tag as5062efe9 --To-tag 695d96db 
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #538
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c: Stopping retransmission on '261090773aaf677807b853f65df7c2cd@10.10.0.15:5060' of Request 102: Match Found
[Mar 29 20:35:28] DEBUG[30640] chan_sip.c: Destroying SIP dialog 261090773aaf677807b853f65df7c2cd@10.10.0.15:5060
[Mar 29 20:35:28] VERBOSE[30640] chan_sip.c: Really destroying SIP dialog '261090773aaf677807b853f65df7c2cd@10.10.0.15:5060' Method: OPTIONS
[Mar 29 20:35:30] DEBUG[31631] pbx.c: Launching 'Playback'
[Mar 29 20:35:30] VERBOSE[31631] pbx.c:     -- Executing [s@from-trunk:4] Playback("SIP/from-trunk-0000000d", "ss-noservice") in new stack

Statistics : Posted by duxy786 • on Thu Mar 28, 2013 5:26 pm • Replies 5 • Views 72

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I don't see any failures in that trace, but as I said, it is a pain to look at them in detail. You need to identify where you think it is going wrong, so that it is possible to concentrate on the problem area.

Statistics : Posted by duxy786 • on Thu Mar 28, 2013 5:26 pm • Replies 5 • Views 72

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Hi David,

Thank you for your reply.

As requested, here is the Asterisk output for a call. I cleared the full log, then did the test. The first call went thought (surprisingly) then the second call failed as expected. The output:

Again same problem when I post the log here, I get the message "Your message contains 277503 characters. The maximum number of allowed characters is 60000.", even within a code block. Can I (you) remove this limitation?

Pastebin link of Asterisk full log: http://pastebin.com/DwA1Hgvu

Duxy786

Statistics : Posted by duxy786 • on Thu Mar 28, 2013 5:26 pm • Replies 5 • Views 72

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Please provide the equivalent information from Asterisk (enable full log, and do:

core set verbose 5
core set debug 5
sip set debug on)

Also, please provide it inline using Code, as having to go to another site every time is a hassle.

Finding information in this sort of log is difficult enough when one is familiar with the format, but much worse if it is from the foreign system.

Statistics : Posted by duxy786 • on Thu Mar 28, 2013 5:26 pm • Replies 5 • Views 72

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