[asterisk-users] Trunk issue

Wed Apr 23 18:20:42 CDT 2014

Are you using freeswitch, or just plain asterisk?  I just setup a trunk between Asterisk and CM this morning, and it works great.... providing that you allow for anonymous calls.

-----Original Message-----
From: "Haley,Scott A" <scott.haley at edwardjones.com>
Sent: Wednesday, April 23, 2014 9:36am
To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com>
Subject: [asterisk-users] Trunk issue

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   http://lists.digium.com/mailman/listinfo/asterisk-usersI have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong?

nxdasterisk-2*CLI>
[Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted
Audio is at 18380
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100003 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.175.135:5060:
INVITE sip:913145152244 at 192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244 at 192.168.175.135>
Contact: <sip:3145152000 at 192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 23 Apr 2014 13:20:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 100 Trying
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
CSeq: 102 INVITE
From: Edward Jones <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244 at 192.168.175.135>
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.175.135:5060 --->
INVITE sip:913145152244 at devjones.com SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:192.168.122.57;lr;phase=terminating>
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Date: Wed, 23 Apr 2014 13:20:59 GMT
Contact: <sip:3145152000 at 192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68>
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Record-Route: <sip:2ca13a6d at 192.168.175.135;transport=udp;lr>
Record-Route: <sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr>
Record-Route: <sip:2ca13a6d at 192.168.175.135;transport=udp;lr>
P-Charging-Vector: icid-value="d13ae820-caef-11e3-9b9c-6c3be5a59e68"
User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: Edward Jones <sip:3145152000 at devjones.com>
From: Edward Jones <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244 at 192.168.175.135>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
Max-Forwards: 66
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 229
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
P-Location: SM;origlocname="Asterisk-2";origsiglocname="Asterisk-2";origmedialocname="Asterisk-2";termlocname="Asterisk-2";termsiglocname="Asterisk-2";smaccounting="true"

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (27 headers 11 lines) ---
Sending to 192.168.175.135:5060 (no NAT)
Sending to 192.168.175.135:5060 (no NAT)
Using INVITE request as basis request - 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
Found peer 'SMtrunk' for '3145152000' from 192.168.175.135:5060
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|g722), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.122.57:18380
Looking for 913145152244 in from-pstn (domain devjones.com)

<--- Reliably Transmitting (no NAT) to 192.168.175.135:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.175.135;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4;received=192.168.175.135;rport=5060
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
From: Edward Jones <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244 at 192.168.175.135>;tag=as119fde8b
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[Apr 23 08:20:59] NOTICE[19026][C-00000003]: chan_sip.c:25450 handle_request_invite: Call from 'SMtrunk' (192.168.175.135:5060) to extension '913145152244' rejected because extension not found in context 'from-pstn'.
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.175.135:5060 --->
ACK sip:913145152244 at devjones.com SIP/2.0
Route: <sip:192.168.122.57;lr;phase=terminating>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
From: Edward Jones <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244 at 192.168.175.135>;tag=as119fde8b
Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
CSeq: 102 ACK
Max-Forwards: 66
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060' Method: ACK

<--- SIP read from UDP:192.168.175.135:5060 --->
SIP/2.0 403 Forbidden (Denial 1732)
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
Server: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.3.1.0.631004
Warning: 399 192.168.175.252 "Restricted Access"
To: <sip:913145152244 at 192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
From: Edward Jones <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.175.135:5060:
ACK sip:913145152244 at 192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: "Edward Jones" <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f
To: <sip:913145152244 at 192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00
Contact: <sip:3145152000 at 192.168.122.57:5060>
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0

---
[Apr 23 08:20:59] WARNING[19026][C-00000002]: chan_sip.c:22991 handle_response_invite: Received response: "Forbidden" from '"Edward Jones" <sip:3145152000 at 192.168.122.57>;tag=as4eecf94f'
Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60 at 192.168.122.57:5060' in 32000 ms (Method: INVITE)
[Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (1) Hangup
[Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:392 attempt_thread: Queued call to SIP/SMtrunk/913145152244 expired without completion after 0 attempts

Thanks,
Scott Haley
IS Voice Projects Team
Edward Jones Investments
Phone: 314-515-2244
Email: scott.haley at edwardjones.com<mailto:scott.haley at edwardjones.com>

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