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[HELP]:PJSIP INVITE hanging at 100-Trying with UK Provider

December 11, 2015, 2:49 am
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Thanks David,

(sorry for the delay posting this: I tried to do it yesterday but the forum was messing me around).

That was my initial thought too, so I stripped out all complicated logic from the target context:
Code: dialplan show | grep -A 5 \'from-voipfone\'
[ Context 'from-voipfone' created by 'pbx_ael' ]
  '_.' =>           1. Answer()                                   [pbx_ael]
                    2. NoOp(Grabbed Call)                         [pbx_ael]
                    3. Goto(bt-in,777799,1)                       [pbx_ael]


I also thought that it might not marry up to this context, hence I tried changing it to fubar-from-voipfone and the same trace would then throw errors.

Even with Verbose on 5 there is no output from the dialplan to follow, hence my impression that this was a PJSIP issue.

Could there be anything funny about AEL-generated contexts? I may try this later on to review.

Input and opinions (even theories about the moon landings at this stage) very welcome!

Statistics : Posted by ijh_uk • on Fri Dec 11, 2015 4:49 am • Replies 7 • Views 755
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[HELP]:PJSIP INVITE hanging at 100-Trying with UK Provider

December 11, 2015, 2:49 am
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Hi,

I have got very frustrated trying to get PJSIP to answer incoming calls from a UK VoIP provider Voipfone. They use a single IP and supply no authentication information on calls (unsurprisingly) and we have used them with chan_sip for years but would like to migrate to PJSIP for future support and to take advantage of some of the transport facilities etc.

Asterisk is a freshly compiled Certified 13.1 sat behind a router with ports 5065 forwarded as a SIP port and 30000-39999 forwarded as RTP range (matching the configuration in asterisk).

We have stripped the config and dialplan right down to basics to test, but calls come in, get a 100 Trying response and never progress to 200 status or hit the dialplan (I have tested that it is not dialplan by changing to a fubar context which causes the call to fail immediately), they just timeout and cancel. I have compared to previous chan_sip traces but do not have a valid PJSIP one to compare to and cannot see anything obvious (other than it not, y'know, working).

Any help or clues greatly appreciated!

/etc/asterisk/pjsip.conf:
Code: *** This Conf has been Sanitised to alias: ***
Public IPs, used in this log: wan1.our.domain=1.2.3.4 wan2.our.domain=5.6.7.8
Private IPs, subnets are show as 192.168.x.y, key local subnet is 192.168.2.y
Voipfone identifiers: aliased to 30000000, voipfone ids partially masked
Phone numbers masked to almost all 7s
*** Generated Fri Dec 11 10:38:00 GMT 2015 ***

[TRANSwan1]
type=transport
protocol=udp
bind=192.168.2.30:5065
local_net=192.168.2.0/23
external_media_address=1.2.3.4
external_signaling_address=1.2.3.4
[AUTHvoipfonePrimary]
type=auth
auth_type=userpass
password=111111
username=30000000*200
[voipfonePrimary]
type=endpoint
aors=AORvoipfonePrimary
transport=TRANSwan1
outbound_auth=AUTHvoipfonePrimary
from_domain=wan1.our.domain
context=from-voipfone
disallow=all
allow=alaw
media_use_received_transport=yes
[AORvoipfonePrimary]
type=aor
contact=sip:195.189.173.27:5060
qualify_frequency=0
[IDvoipfone]
type=identify
endpoint=voipfonePrimary
match=195.189.172.1/23
[REGvoipfonePrimary]
type=registration
outbound_auth=AUTHvoipfonePrimary
server_uri=sip:30000000*200@sip.voipfone.net
client_uri=sip:30000000*200@wan1.our.domain:5065
contact_user=30000000*200
auth_rejection_permanent=no
transport=TRANSwan1
retry_interval=60
expiration=300


/var/log/asterisk/full (debug=5,verbose=5,pjsip logger on)
Code: *** This Log has been Sanitised to alias: ***
Public IPs, used in this log: wan1.our.domain=1.2.3.4 wan2.our.domain=5.6.7.8
Private IPs, subnets are show as 192.168.x.y, key local subnet is 192.168.2.y
Voipfone identifiers: aliased to 30000000, voipfone ids partially masked
Phone numbers masked to almost all 7s
*** Generated Wed Dec  9 18:34:15 GMT 2015 ***

[2015-12-09 17:36:38.3] DEBUG[6067] pjsip:    sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=102 (rdata0x22ef668)
[2015-12-09 17:36:38.3] VERBOSE[6067] res_pjsip_logger.c: <--- Received SIP request (954 bytes) from UDP:195.189.173.27:5060 --->
INVITE sip:30000000*200@1.2.3.4:5065;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1 SIP/2.0
Via: SIP/2.0/UDP 195.189.173.27:5060;branch=z9hG4bK638b6901;rport
From: "Main" <sip:07777777789@195.189.173.27>;tag=VFaaaaaaaace8a35a8431a34d370b1
To: <sip:30000000*200@1.2.3.4:5065;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1>
Contact: <sip:07777777789@195.189.173.27>
Call-ID: VFaaaaaaaa93914b5830f1e55a826c3b@voipfone
CSeq: 102 INVITE
User-Agent: Voipfone Sip Network
Date: Wed, 09 Dec 2015 17:36:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 337

v=0
o=root 39459 39459 IN IP4 195.189.173.27
s=session
c=IN IP4 195.189.173.27
t=0 0
m=audio 55154 RTP/AVP 8 2 97 3 110 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    sip_endpoint.c Distributing rdata to modules: Request msg INVITE/cseq=102 (rdata0x22556e8)
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_endpoint_identifier_user.c: Could not identify endpoint by username '07777777789'
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: Splitting '195.189.173.27' into...
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: ...host '195.189.173.27' and port ''.
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_endpoint_identifier_ip.c: Source address 195.189.173.27:5060 matches identify 'IDvoipfone'
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_endpoint_identifier_ip.c: Retrieved endpoint voipfonePrimary
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    tsx0x7f3468007 ..Transaction created for Request msg INVITE/cseq=102 (rdata0x22556e8)
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    tsx0x7f3468007 .Incoming Request msg INVITE/cseq=102 (rdata0x22556e8) in state Null
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    tsx0x7f3468007 ..State changed from Null to Trying, event=RX_MSG
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e ...Transaction tsx0x7f3468007268 state changed to Trying
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e .UAS dialog created
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e .Module mod-invite added as dialog usage, data=0x7f346801bcd8
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e ..Session count inc to 2 by mod-invite
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    inv0x7f346800e .UAS invite session created for dialog dlg0x7f346800ebf8
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e .Module Session Module added as dialog usage, data=(nil)
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e ..Session count inc to 2 by Session Module
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Negotiating incoming SDP media stream 'audio' using audio SDP handler
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: Splitting '195.189.173.27' into...
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: ...host '195.189.173.27' and port ''.
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f346800a278'
[2015-12-09 17:36:38.3] DEBUG[6062] res_rtp_asterisk.c: Allocated port 34220 for RTP instance '0x7f346800a278'
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    icess0x7f34680 ICE session created, comp_cnt=2, role is Unknown agent
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: Splitting '192.168.2.30' into...
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: ...host '192.168.2.30' and port ''.
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    icess0x7f34680 Candidate 0 added: comp_id=1, type=host, foundation=Hc0a8ca1e, addr=192.168.2.30:34220, base=192.168.2.30:34220, prio=0x7effffff (2130706431)
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: RTP instance '0x7f346800a278' is setup and ready to go
[2015-12-09 17:36:38.3] DEBUG[6062] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f346800a278'
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: Splitting '192.168.2.30' into...
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: ...host '192.168.2.30' and port ''.
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    icess0x7f34680 Candidate 1 added: comp_id=2, type=host, foundation=Hc0a8ca1e, addr=192.168.2.30:34221, base=192.168.2.30:34221, prio=0x7efffffe (2130706430)
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    icess0x7f34680 Destroying ICE session 0x7f3468021058
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    stuse0x7f34680 STUN session 0x7f3468012298 destroy request, ref_cnt=4
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    stuse0x7f34680 STUN session 0x7f3468014a78 destroy request, ref_cnt=3
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:     ice_session.c ICE session 0x7f3468021058 destroyed
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    stun_session.c STUN session 0x7f3468012298 destroyed
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    stun_session.c STUN session 0x7f3468014a78 destroyed
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Setting payload 8 (0x7f346800b668) based on m type on 0x7f3483df1580
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Setting payload 2 (0x7f346800b668) based on m type on 0x7f3483df1580
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Setting payload 97 (0x7f346800b668) based on m type on 0x7f3483df1580
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Setting payload 3 (0x7f346800b668) based on m type on 0x7f3483df1580
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Setting payload 110 (0x7f346800b668) based on m type on 0x7f3483df1580
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Setting payload 101 (0x7f346800b668) based on m type on 0x7f3483df1580
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Copying payload 2 (0x7f3468006778) from 0x7f3483df1580 to 0x7f346800a440
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Copying payload 3 (0x7f34680171d8) from 0x7f3483df1580 to 0x7f346800a440
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Copying payload 8 (0x7f346800b6e8) from 0x7f3483df1580 to 0x7f346800a440
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Copying payload 97 (0x7f346800b768) from 0x7f3483df1580 to 0x7f346800a440
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Copying payload 101 (0x7f346802add8) from 0x7f3483df1580 to 0x7f346800a440
[2015-12-09 17:36:38.3] DEBUG[6062] rtp_engine.c: Copying payload 110 (0x7f346802ad58) from 0x7f3483df1580 to 0x7f346800a440
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Media stream 'audio' handled by audio
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:          endpoint .Response msg 100/INVITE/cseq=102 (tdta0x7f3468005200) created
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e .Initial answer Response msg 100/INVITE/cseq=102 (tdta0x7f3468005200)
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Method is INVITE, Response is 100 Trying
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    inv0x7f346800e .Sending Response msg 100/INVITE/cseq=102 (tdta0x7f3468005200)
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e ..Sending Response msg 100/INVITE/cseq=102 (tdta0x7f3468005200)
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    tsx0x7f3468007 ..Sending Response msg 100/INVITE/cseq=102 (tdta0x7f3468005200) in state Trying
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: Splitting '195.189.173.27' into...
[2015-12-09 17:36:38.3] DEBUG[6062] netsock2.c: ...host '195.189.173.27' and port ''.
[2015-12-09 17:36:38.3] VERBOSE[6062] res_pjsip_logger.c: <--- Transmitting SIP response (371 bytes) to UDP:195.189.173.27:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.189.173.27:5060;rport=5060;received=195.189.173.27;branch=z9hG4bK638b6901
Call-ID: VFaaaaaaaa93914b5830f1e55a826c3b@voipfone
From: "Main" <sip:07777777789@195.189.173.27>;tag=VFaaaaaaaace8a35a8431a34d370b1
To: <sip:30000000*200@1.2.3.4;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1>
CSeq: 102 INVITE
Content-Length:  0


[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    tsx0x7f3468007 ...State changed from Trying to Proceeding, event=TX_MSG
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e ....Transaction tsx0x7f3468007268 state changed to Proceeding
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The state change pertains to the session with voipfonePrimary
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f3468007268)
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: There is no transaction involved in this state change
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The current inv state is INCOMING
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Source of transaction state change is TX_MSG
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Sending response
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Method is INVITE, Response is 100 Trying
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The state change pertains to the session with voipfonePrimary
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f3468007268)
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The transaction involved in this state change is 0x7f3468007268
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The current transaction state is Proceeding
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The transaction state change event is TX_MSG
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: The current inv state is INCOMING
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Sending response
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Method is INVITE, Response is 100 Trying
[2015-12-09 17:36:38.3] DEBUG[6062] res_pjsip_session.c: Method is INVITE
[2015-12-09 17:36:38.3] DEBUG[6062] pjsip:    dlg0x7f346800e Module NAT added as dialog usage, data=(nil)

<<<REMOVED Successful REGISTER dialog for clarity>>>

[2015-12-09 17:37:00.3] DEBUG[6067] pjsip:    sip_endpoint.c Processing incoming message: Request msg CANCEL/cseq=102 (rdata0x22ef668)
[2015-12-09 17:37:00.3] VERBOSE[6067] res_pjsip_logger.c: <--- Received SIP request (498 bytes) from UDP:195.189.173.27:5060 --->
CANCEL sip:30000000*200@1.2.3.4:5065;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1 SIP/2.0
Via: SIP/2.0/UDP 195.189.173.27:5060;branch=z9hG4bK638b6901;rport
From: "Main" <sip:07777777789@195.189.173.27>;tag=VFaaaaaaaace8a35a8431a34d370b1
To: <sip:30000000*200@1.2.3.4:5065;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1>
Contact: <sip:07777777789@195.189.173.27>
Call-ID: VFaaaaaaaa93914b5830f1e55a826c3b@voipfone
CSeq: 102 CANCEL
User-Agent: Voipfone Sip Network
Content-Length: 0


[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    sip_endpoint.c Distributing rdata to modules: Request msg CANCEL/cseq=102 (rdata0x22556e8)
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e .Received Request msg CANCEL/cseq=102 (rdata0x22556e8)
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f346801b ...Transaction created for Request msg CANCEL/cseq=102 (rdata0x22556e8)
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f346801b ..Incoming Request msg CANCEL/cseq=102 (rdata0x22556e8) in state Null
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f346801b ...State changed from Null to Trying, event=RX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ....Transaction tsx0x7f346801bf68 state changed to Trying
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The state change pertains to the session with voipfonePrimary
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f3468007268)
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction involved in this state change is 0x7f346801bf68
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current transaction state is Trying
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction state change event is RX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current inv state is INCOMING
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Received request
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Method is CANCEL
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:          endpoint .....Response msg 200/CANCEL/cseq=102 (tdta0x7f3468018f60) created
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ......Sending Response msg 200/CANCEL/cseq=102 (tdta0x7f3468018f60)
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f346801b ......Sending Response msg 200/CANCEL/cseq=102 (tdta0x7f3468018f60) in state Trying
[2015-12-09 17:37:00.3] DEBUG[6062] netsock2.c: Splitting '195.189.173.27' into...
[2015-12-09 17:37:00.3] DEBUG[6062] netsock2.c: ...host '195.189.173.27' and port ''.
[2015-12-09 17:37:00.3] VERBOSE[6062] res_pjsip_logger.c: <--- Transmitting SIP response (408 bytes) to UDP:195.189.173.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.189.173.27:5060;rport=5060;received=195.189.173.27;branch=z9hG4bK638b6901
Call-ID: VFaaaaaaaa93914b5830f1e55a826c3b@voipfone
From: "Main" <sip:07777777789@195.189.173.27>;tag=VFaaaaaaaace8a35a8431a34d370b1
To: <sip:30000000*200@1.2.3.4;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1>;tag=80d4f907-97d9-4f60-aca9-e31e85c36335
CSeq: 102 CANCEL
Content-Length:  0


[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f346801b .......State changed from Trying to Completed, event=TX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ........Transaction tsx0x7f346801bf68 state changed to Completed
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The state change pertains to the session with voipfonePrimary
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f3468007268)
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction involved in this state change is 0x7f346801bf68
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current transaction state is Completed
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction state change event is TX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current inv state is INCOMING
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Sending response
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Method is CANCEL, Response is 200 OK
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ......Sending Response msg 487/INVITE/cseq=102 (tdta0x7f3468005200)
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f3468007 ......Sending Response msg 487/INVITE/cseq=102 (tdta0x7f3468005200) in state Proceeding
[2015-12-09 17:37:00.3] DEBUG[6062] netsock2.c: Splitting '195.189.173.27' into...
[2015-12-09 17:37:00.3] DEBUG[6062] netsock2.c: ...host '195.189.173.27' and port ''.
[2015-12-09 17:37:00.3] VERBOSE[6062] res_pjsip_logger.c: <--- Transmitting SIP response (424 bytes) to UDP:195.189.173.27:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 195.189.173.27:5060;rport=5060;received=195.189.173.27;branch=z9hG4bK638b6901
Call-ID: VFaaaaaaaa93914b5830f1e55a826c3b@voipfone
From: "Main" <sip:07777777789@195.189.173.27>;tag=VFaaaaaaaace8a35a8431a34d370b1
To: <sip:30000000*200@1.2.3.4;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1>;tag=80d4f907-97d9-4f60-aca9-e31e85c36335
CSeq: 102 INVITE
Content-Length:  0


[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f3468007 .......State changed from Proceeding to Completed, event=TX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ........Transaction tsx0x7f3468007268 state changed to Completed
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The state change pertains to the session with voipfonePrimary
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f3468007268)
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: There is no transaction involved in this state change
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current inv state is DISCONNCTD
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Source of transaction state change is TX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Sending response
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Method is INVITE, Response is 487 Request Terminated
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Destroying SIP session with endpoint voipfonePrimary
[2015-12-09 17:37:00.3] DEBUG[6062] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f346800a278'
[2015-12-09 17:37:00.3] DEBUG[6062] rtp_engine.c: Destroyed RTP instance '0x7f346800a278'
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ..........Session count dec to 6 by Session Module
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ..........Session count dec to 5 by mod-invite
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: inv_session 0x7f346801bcd8 has no ast session
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f3468007268)
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction involved in this state change is 0x7f3468007268
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current transaction state is Completed
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction state change event is TX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current inv state is DISCONNCTD
[2015-12-09 17:37:00.3] DEBUG[6062] taskprocessor.c: destroying taskprocessor '8e62eeb2-cd3e-4da5-8c79-ee3862217338'
[2015-12-09 17:37:00.3] DEBUG[6067] pjsip:    sip_endpoint.c Processing incoming message: Request msg ACK/cseq=102 (rdata0x22ef668)
[2015-12-09 17:37:00.3] VERBOSE[6067] res_pjsip_logger.c: <--- Received SIP request (533 bytes) from UDP:195.189.173.27:5060 --->
ACK sip:30000000*200@1.2.3.4:5065;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1 SIP/2.0
Via: SIP/2.0/UDP 195.189.173.27:5060;branch=z9hG4bK638b6901;rport
From: "Main" <sip:07777777789@195.189.173.27>;tag=VFaaaaaaaace8a35a8431a34d370b1
To: <sip:30000000*200@1.2.3.4:5065;x-uid=aaaaaaaa-ff2c-4857-bc79-4b1dc52855c1>;tag=80d4f907-97d9-4f60-aca9-e31e85c36335
Contact: <sip:07777777789@195.189.173.27>
Call-ID: VFaaaaaaaa93914b5830f1e55a826c3b@voipfone
CSeq: 102 ACK
User-Agent: Voipfone Sip Network
Content-Length: 0


[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    sip_endpoint.c Distributing rdata to modules: Request msg ACK/cseq=102 (rdata0x22556e8)
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_endpoint_identifier_user.c: Could not identify endpoint by username '07777777789'
[2015-12-09 17:37:00.3] DEBUG[6062] netsock2.c: Splitting '195.189.173.27' into...
[2015-12-09 17:37:00.3] DEBUG[6062] netsock2.c: ...host '195.189.173.27' and port ''.
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_endpoint_identifier_ip.c: Source address 195.189.173.27:5060 matches identify 'IDvoipfone'
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_endpoint_identifier_ip.c: Retrieved endpoint voipfonePrimary
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f3468007 .Incoming Request msg ACK/cseq=102 (rdata0x22556e8) in state Completed
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    tsx0x7f3468007 ..State changed from Completed to Confirmed, event=RX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] pjsip:    dlg0x7f346800e ...Transaction tsx0x7f3468007268 state changed to Confirmed
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: inv_session 0x7f346801bcd8 has no ast session
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f3468007268)
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction involved in this state change is 0x7f3468007268
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current transaction state is Confirmed
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The transaction state change event is RX_MSG
[2015-12-09 17:37:00.3] DEBUG[6062] res_pjsip_session.c: The current inv state is DISCONNCTD
[2015-12-09 17:37:00.4] DEBUG[6067] pjsip:    tsx0x7f3468007 Timeout timer event
[2015-12-09 17:37:00.4] DEBUG[6067] pjsip:    tsx0x7f3468007 .State changed from Confirmed to Terminated, event=TIMER
[2015-12-09 17:37:00.4] DEBUG[6067] pjsip:    dlg0x7f346800e ..Transaction tsx0x7f3468007268 state changed to Terminated
[2015-12-09 17:37:00.4] DEBUG[6067] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2015-12-09 17:37:00.4] DEBUG[6067] res_pjsip_session.c: inv_session 0x7f346801bcd8 has no ast session
[2015-12-09 17:37:00.4] DEBUG[6067] res_pjsip_session.c: The inv session does NOT have an invite_tsx
[2015-12-09 17:37:00.4] DEBUG[6067] res_pjsip_session.c: The transaction involved in this state change is 0x7f3468007268
[2015-12-09 17:37:00.4] DEBUG[6067] res_pjsip_session.c: The current transaction state is Terminated
[2015-12-09 17:37:00.4] DEBUG[6067] res_pjsip_session.c: The transaction state change event is TIMER
[2015-12-09 17:37:00.4] DEBUG[6067] res_pjsip_session.c: The current inv state is DISCONNCTD
[2015-12-09 17:37:00.4] DEBUG[6067] pjsip:    tsx0x7f3468007 Timeout timer event
[2015-12-09 17:37:00.4] DEBUG[6067] pjsip:    tsx0x7f3468007 .State changed from Terminated to Destroyed, event=TIMER
[2015-12-09 17:37:00.4] DEBUG[6067] pjsip:    tdta0x7f346800 ..Destroying txdata Response msg 487/INVITE/cseq=102 (tdta0x7f3468005200)
[2015-12-09 17:37:00.4] DEBUG[6067] pjsip:    tsx0x7f3468007 Transaction destroyed!
[2015-12-09 17:37:04.8] VERBOSE[6132] asterisk.c: Remote UNIX connection disconnected


Statistics : Posted by ijh_uk • on Fri Dec 11, 2015 4:49 am • Replies 7 • Views 755
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[HELP]:PJSIP INVITE hanging at 100-Trying with UK Provider

December 11, 2015, 2:49 am
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You won't get beyond trying unless the dialplan does something that causes this. You don't appear t have verbose logging on to show what the diaplan is doing, and you haven't provided the dialplan.

Statistics : Posted by ijh_uk • on Fri Dec 11, 2015 4:49 am • Replies 7 • Views 755
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[HELP]:PJSIP INVITE hanging at 100-Trying with UK Provider

December 11, 2015, 2:49 am
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Further piece of information from a couple of people on the provider's forums (which generally are better known for their tumbleweed than anything else!)

A user has reported the same issue, which he eventually traced to failing to menuselect chan_pjsip (i.e. not actually compiling it). As best I can tell that is not my direct issue, but I assume this fits into your line of enquiry on the compilation and the PJSIP linkage. Feeling out of my depth now!

Statistics : Posted by ijh_uk • on Fri Dec 11, 2015 4:49 am • Replies 8 • Views 840
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Reachable and Unreachable

January 2, 2016, 8:20 am
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You have qualify enabled.

For unreachable, there was no response within the value of the qualify parameter when asterisk sent an OPTIONS request to the peer. For reachable, the peer was previous unreachable but there was a response within the time defined by qualify for the last OPTIONS sent. (It is pssible that unreachable needs more than one consecutive failure.

The peer decides what port it is on for host=dynamic.

Statistics : Posted by geraldinejns • on Sat Jan 02, 2016 10:20 am • Replies 1 • Views 13
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Reachable and Unreachable

January 2, 2016, 8:20 am
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Does anyone have a clue as to why the below message happens. One minute i can dial out and the next minute i can't.

NOTICE[1886]: chan_sip.c:23571 handle_response_peerpoke: Peer '203' is now Reachable. (61ms/2000ms)

NOTICE[1886]: chan_sip.c:29504 sip_poke_noanswer: Peer '203' is now UNREACHABLE! Last qualify:61

Also, when i type sip show peers,

203 123.456.789.101 D Yes A 5060 OK
202 123.456.789.101 D Yes A 2581 UNREACHABLE
201 123.456.789.101 D Yes A 2582 OK
204 123.456.789.101 D Yes A 5060 UNREACHABLE

aren't they suppose to all be on the same port: 5060?

How can i tell if i'm being hacked?

Statistics : Posted by geraldinejns • on Sat Jan 02, 2016 10:20 am • Replies 1 • Views 13
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WebRTC (SipML5) on Doubango registers but media fails.

January 2, 2016, 3:47 pm
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"Media stream permission denied" is because

One major change is WebRTC won’t work without HTTPS.
According internal security policy Chrome browser does not support getUserMedia() for unsecure pages since version 47. So you will not be able to use microphone if your page is not HTTPS.

http://flashphoner.com/getusermedia-no- ... e-origins/

Statistics : Posted by lardconcepts • on Sat Jan 02, 2016 5:47 pm • Replies 1 • Views 80
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WebRTC (SipML5) on Doubango registers but media fails.

January 2, 2016, 3:47 pm
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Asterisk 13.6 - Chrome 48.0.2564.48 beta-m (64-bit)

Following the instructions at https://wiki.asterisk.org/wiki/display/ ... ing+SIPML5, somehow I managed to get the Douabango SipML5 demo webpage to register and login to my server. But when it comes to making a call, I get the following errors:

In the browser console, it seems to be trying...
Code: GET http://www.doubango.org/sipml5/null
Request Method:GET
Status Code:404 Not Found

and by the dialler, the following message flashes up
Quote:"Media stream permission denied"

One thing to note - I had to change https://www.doubango.org/sipml5/call.htm to http://www.doubango.org/sipml5/call.htm otherwise it was complaining about me attempting an insecure call. I might be being thick here, but I can't find how to configure the wss 8089 port.

That said, once I changed to http, it definitely registers just fine on my Asterisk.

Before I keep on too late into the night, does that "404 not found" error look like it might be something at their end? I don't want to start trying to debug a problem that's somewhere else.

By the way, when the call is attempted, nothing flashes by in the pjsip log, so it's not even hitting my server, unlike the registration which I can see going just fine Image

Statistics : Posted by lardconcepts • on Sat Jan 02, 2016 5:47 pm • Replies 1 • Views 80
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Search

Please check my configuration

December 31, 2015, 1:45 am
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A workaround for the caller id would be set the option sendrpid=yes on your sipgate config, canreinvite was renamed to directmedia, RTP start and RTP end are configured on the rtp.conf file

Statistics : Posted by dogman • on Thu Dec 31, 2015 3:45 am • Replies 2 • Views 230
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Please check my configuration

December 31, 2015, 1:45 am
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Caller ID won't work because you are using fromuser and haven't provided any alternative way of signalling caller ID (this assumes that sipgate will trust any caller IDs from you).

canreinvite is deprecated and may even be obsolete.

nat=yes is deprecated. You should set the options that you actually need, if any.

type=friend is not necessary, and is, therefore, a security risk; use type=peer.

For a typical DMZ, there will be no NAT to the inside, so localnet should include the inside subnets.

There should be no bits set before the / in localnet that are not set after it (may or may not cause a problem).

All the codec choices should be the same to avoid unnecessary transcoding and a slight audio degredation. Europe use A-law, so if sipgate breaks out to the PSTN in the EU, you should use alaw.

allowguest should be explicitly set to "no".

There should not normally be any "s in sip.conf.

Although often done, making sip.conf names easily guessable, by using extension numbers, is bad security practice.

remotesecret=<password> more clearly indicates what it does than using insecure=invite.

username is a deprecated parameter name and serves no useful purpose for internal devices.

Statistics : Posted by dogman • on Thu Dec 31, 2015 3:45 am • Replies 2 • Views 230
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Please check my configuration

December 31, 2015, 1:45 am
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Hi all, I'm relatively new to Asterisk and I just need some help making sure my configuration is optimal. I do not believe I have everything correct.

My phone system is in a small business (don't worry they aren't paying me), about 20 hardphones (Cisco 7960). They have a SIP trunk to sipgate UK which is used for ingoing and outgoing calls. As they are using a basic business broadband connection with a crap router and behind NAT, the PBX is in DMZ to avoid one way audio problems for now until I set them up with a proper pfsense firewall configured for SIP. I have moved SIP onto a non-standard port as a basic security measure temporarily.

Each phone has its own internal 3-digit extension starting with 2 (201, 202, 203 etc). All of these can call each other.
Certain phones have a second line with extension 301, which the inbound SIP trunk calls are routed to.
All phones can call out via SIP trunk by dialling 9 before the number.
50000 calls the sipgate voicemail from any phone via SIP trunk.
100 calls the sipgate test number from any phone via SIP trunk.
299 calls an internal test number which has an echo test.

Passwords etc removed from following configs
sip.conf:

Quote:[general]
context=default
allowoverlap=no
bindport=5061; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes
mwi_from=asterisk
;allowguest=no
alwaysauthreject=yes
localnet=192.168.0.109/255.255.255.0
rtpstart=10000
rtpend=10200

register => sipgate-id:sipgate-pass@sipgate.co.uk/sipgate-id


[sipgate]
type=peer
secret=sipgate-pass
insecure=invite
username=sipgate-id
defaultuser=sipgate-id
fromuser=sipgate-id
context=sipgate-in
fromdomain=sipgate.co.uk
host=sipgate.co.uk
outboundproxy=proxy.live.sipgate.co.uk
nat=yes
qualify=yes
disallow=all
allow=alaw
dtmfmode=rfc2833

[201]
type=friend
host=dynamic
dtmfmode=rfc2833
nat=yes
username=201
secret=pass
context=internal
canreinvite=no
callerid="Phone 1"

[202]
type=friend
host=dynamic
dtmfmode=rfc2833
nat=yes
disallow=all
allow=ulaw
username=202
secret=pass
context=internal
canreinvite=no
callerid="Phone 2"

[203]
type=friend
host=dynamic
dtmfmode=rfc2833
nat=yes
disallow=all
allow=ulaw
username=203
secret=pass
context=internal
canreinvite=no
callerid="Phone 3"

~

[301]
type=friend
host=dynamic
dtmfmode=rfc2833
nat=yes
disallow=all
allow=ulaw
username=301
secret=pass
context=sipgate-out
canreinvite=no
callerid="External"


extensions.conf:

Quote:[default]
exten => i,1,Hangup

;route sipgate inbound calls to ext 301
exten => sipgate-id,n,Dial(SIP/301)
exten => sipgate-id,n,Hangup

[internal]
;dial 9 to route any number via sipgate
exten => _9[0-9].,1,Set(CALLERID(num)=sipgate-id)
exten => _9[0-9].,n,Dial(SIP/${EXTEN:1}@sipgate,25,trg)
exten => _9[0-9].,n,Hangup

;internal extensions below
exten => 201,1,Dial(SIP/201,25)
exten => 201,n,Hangup

exten => 202,1,Dial(SIP/202,25)
exten => 202,n,Hangup

exten => 203,1,Dial(SIP/203,25)
exten => 203,n,Hangup

~

;forward extension 100 to sipgate test number
exten => 100,1,Set(CALLERID(num)=2077413e0)
exten => 100,n,Dial(SIP/10000@sipgate,30,trg)
exten => 100,n,Hangup

;internal test number
exten => 299,1,Ringing()
exten => 299,n,Wait(3)
exten => 299,n,Answer()
exten => 299,n,Playback(dir-multi3)
exten => 299,n,SayDigits(${CALLERID(num)})
exten => 299,n,Wait(1)
exten => 299,n,Playback(channel)
exten => 299,n,Wait(1)
exten => 299,n,SayAlpha(${CHANNEL})
exten => 299,n,Wait(1)
exten => 299,n,Playback(readback-instructions)
exten => 299,n,Record(/tmp/299-${UNIQUEID}.wav,0,30)
exten => 299,n,Playback(/tmp/299-${UNIQUEID})
exten => 299,n,System(rm /tmp/299-${UNIQUEID}.wav)
exten => 299,n,Wait(1)
exten => 299,n,Playback(vm-goodbye)
exten => 299,n,Hangup

;help make the voicemail button call sipgate voicemail
exten => 50000,1,Dial(SIP/50000@sipgate,25)
exten => 50000,n,Hangup



Please advise if there are any blaring errors in this configuration. I'm sure it is wrong, but it's the only way I could get everything to work properly!

Thanks.
dogman

Statistics : Posted by dogman • on Thu Dec 31, 2015 3:45 am • Replies 2 • Views 230
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Please check my configuration

December 31, 2015, 1:45 am
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Thanks for the reply david77. I have made the changes you said, but I have run into some problems. If I set type=peer and remove the nat=yes option, the phones cannot make any calls, internal or through SIP trunk. In asterisk console I see this error:

Quote:[Dec 31 18:07:38] WARNING[32259]: chan_sip.c:8584 check_auth: username mismatch, have <301>, digest has <201>
[Dec 31 18:07:38] NOTICE[32259]: chan_sip.c:14351 handle_request_invite: Failed to authenticate user "Phone 1" <sip:201@192.168.0.109>;tag=000af4a3cc320025138ba437-6760143a


I'm not quite sure what's going on here. If I change back to type=friend, everything works again.

Statistics : Posted by dogman • on Thu Dec 31, 2015 3:45 am • Replies 2 • Views 230
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Trunkmonitor

November 18, 2015, 3:29 am
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Trunkmonitor

November 18, 2015, 3:29 am
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Hi , I have just build asterisk 1.8 test machine on centos 6.i want to monitor all the trunks ,iax,sip and Dadhi.however the agi-bin directoty is empty. cd /var/lib/asterisk/agi-bin

I want to edit the file /var/lib/asterisk/agi-bin/trunkmonitor but is it not there.

thank you in advance

Statistics : Posted by mathabathe • on Wed Nov 18, 2015 5:29 am • Replies 7 • Views 771
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RTP Quality Asterisk 11 vs 13

December 27, 2015, 12:26 am
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HI Joshua,

thank you for your reply.

I'm also confused.
I'm running the latest AsteriskNOW version with a Asterisk 11.20.
Everything is looking brilliant.
After switching only the Asterisk Version to 13.6 the RTP quality is starting to drop.

I'm working for a company which is developing RTP monitoring software and we are inspecting the inter arrival time of the RTP packets. We are running our own software to monitor our VoIP supplier and there I spotted difference between the Asterisk versions.

I'm happy to provide you more details via webex or so if you are interested as the version switch is just one command.

Please feel free to contract me also directly.

Best regards
Christoph

Statistics : Posted by TimmiOrg • on Sun Dec 27, 2015 2:26 am • Replies 2 • Views 213
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RTP Quality Asterisk 11 vs 13

December 27, 2015, 12:26 am
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I've seen no reports elsewhere of any problems and haven't heard (ha) of issues with any deployments I know of. What devices and codecs are in use? Any transcoding going on?

Statistics : Posted by TimmiOrg • on Sun Dec 27, 2015 2:26 am • Replies 2 • Views 213
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RTP Quality Asterisk 11 vs 13

December 27, 2015, 12:26 am
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Hi All,

I have an general question regarding the RTP quality of the Astersik 13. We recently switched our internal PBX to Asterisk 13 and since then we are seeing more Jitter then before.

Yesterday I switched back to Asterisk 11 and the Jitter is gone.

Please note the system is the latestet Asterisk NOW installation and running in a VM. The setting of the VM was and is the same. I just changed the Asterisk version.

Does anyone knows about this topic?

I'm also happy to discuss this in the developer mailing list if you think it is more useful there.

Best regards
Christoph

Statistics : Posted by TimmiOrg • on Sun Dec 27, 2015 2:26 am • Replies 2 • Views 213
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Configuring extensions.conf and sip.conf

January 3, 2016, 10:29 pm
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it seems default is the context used for internal calls, so change the context in your sip peers from out to default

Statistics : Posted by alemono95 • on Mon Jan 04, 2016 12:29 am • Replies 4 • Views 121
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Configuring extensions.conf and sip.conf

January 3, 2016, 10:29 pm
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You have no devices in a context for which extensions are defined in extensions.conf.

autofallthrough=no has been a very unusual setting since autofallthrough was introduced.

In addition:

type=friend is usually a bad idea.

Are the phones really inside NAT and the PABX outside?

Statistics : Posted by alemono95 • on Mon Jan 04, 2016 12:29 am • Replies 4 • Views 121
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Configuring extensions.conf and sip.conf

January 3, 2016, 10:29 pm
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Samples come with the source distribution.

Statistics : Posted by alemono95 • on Mon Jan 04, 2016 12:29 am • Replies 4 • Views 121
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