It is just plain Asterisk. I solved the original problem of it not being in the <from-pstn> context, now I am getting a rejected error I believe from the CM.
Thanks, Scott Haley 5-2244 -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of richard.seg...@marisec.ca Sent: Wednesday, April 23, 2014 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Trunk issue 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.ha...@edwardjones.com> Sent: Wednesday, April 23, 2014 9:36am To: "asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com> Subject: [asterisk-users] Trunk issue -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: 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@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@192.168.122.57>;tag=as4eecf94f To: <sip:913145152244@192.168.175.135> Contact: <sip:3145152000@192.168.122.57:5060> Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@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@192.168.122.57:5060 CSeq: 102 INVITE From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f To: <sip:913145152244@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:913145152...@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@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@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@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:3145152...@devjones.com> From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f To: <sip:913145152244@192.168.175.135> Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@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@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@192.168.122.57>;tag=as4eecf94f To: <sip:913145152244@192.168.175.135>;tag=as119fde8b Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@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@192.168.122.57:5060' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.175.135:5060 ---> ACK sip:913145152...@devjones.com SIP/2.0 Route: <sip:192.168.122.57;lr;phase=terminating> Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f To: <sip:913145152244@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@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@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00 From: Edward Jones <sip:3145152000@192.168.122.57>;tag=as4eecf94f Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@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@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@192.168.122.57>;tag=as4eecf94f To: <sip:913145152244@192.168.175.135>;tag=8072a3b71bcde31d444535cfeab00 Contact: <sip:3145152000@192.168.122.57:5060> Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@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@192.168.122.57>;tag=as4eecf94f' Scheduling destruction of SIP dialog '504b8ce74a81e0f90ba457e77e8c9e60@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.ha...@edwardjones.com<mailto:scott.ha...@edwardjones.com> If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure<http://www.edwardjones.com/US_email_disclosure>. Edward D. Jones & Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
