Agasthian P
2015-Sep-17 16:01 UTC
[asterisk-users] Calls to Ring Group not working. FreePBX.
Hi All, Hi All, I am trying to create an Inbound route destined to a Ring Group through a SIP trunk. I am able to call the extensions directly, but unable to call a Ring Group or an IVR through the Inbound Route config. I am really not sure, what i am missing. When the DID for the IVR or Ring Group is called, getting the message from the Asterisk that "the call cannot be completed, please check your number". I am doing the configuration using FreePBX and the Asterisk version is 12. The Inbound Route configuration for the IVR :- 1. DID Number : 2000 2. Ring Groups : RG<600> SIP Peer details :- host=20.1.1.170 type=friend port=5060 nat=no disallow=all allow=ulaw,alaw qualify=yes canreinvite=yes context=from-trunk When 2000, is dialled, the DID in the SIP Invite is the same, but still getting the error message. SIP Logs :- Invite to the DID 2000 for Ring Group ----> 100 Trying <----- 183 Session Progess <----- (Playing the error message) ------------------------------ <--- SIP read from UDP:20.1.1.170:5060 ---> INVITE sip:2000 at 20.1.1.58:5060 SIP/2.0 Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395 To: Date: Fri, 11 Sep 2015 14:06:41 GMT Call-ID: 52087400-5f21dff1-354b2-aa010114 at 20.1.1.170 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM10.5 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence Supported: X-cisco-srtp-fallback,X-cisco-original-called Cisco-Guid: 1376285696-0000065536-0000002594-2852192532 Session-Expires: 1800 P-Asserted-Identity: Remote-Party-ID: ;party=calling;screen=yes;privacy=off Contact: ;bfcp Max-Forwards: 69 Content-Type: application/sdp Content-Length: 198 v=0 o=CiscoSystemsCCM-SIP 787014 1 IN IP4 20.1.1.170 s=SIP Call c=IN IP4 20.1.1.170 t=0 0 m=audio 25986 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (22 headers 9 lines) --- Sending to 20.1.1.170:5060 (no NAT) Sending to 20.1.1.170:5060 (no NAT) Using INVITE request as basis request - 52087400-5f21dff1-354b2-aa010114 at 20.1.1.170 Found peer '2723' for '2723' from 20.1.1.170:5060 Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 20.1.1.170:25986 Looking for 2000 in from-internal (domain 20.1.1.58) list_route: hop: <--- Transmitting (NAT) to 20.1.1.170:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 20.1.1.170:5060 ;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060 From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395 To: Call-ID: 52087400-5f21dff1-354b2-aa010114 at 20.1.1.170 CSeq: 101 INVITE Server: FPBX-12.0.76(11.19.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 16598 Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 20.1.1.170:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 20.1.1.170:5060 ;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060 From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395 To: ;tag=as3e6a1653 Call-ID: 52087400-5f21dff1-354b2-aa010114 at 20.1.1.170 CSeq: 101 INVITE Server: FPBX-12.0.76(11.19.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 228 v=0 o=root 881046367 881046367 IN IP4 20.1.1.58 s=Asterisk PBX 11.19.0 c=IN IP4 20.1.1.58 t=0 0 m=audio 16598 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ==================================================== Anything i am missing here ? Also please let me know, if you need any other logs to help me in this. Thanks a lot ! Agasthian P -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150917/c83b6210/attachment.html>