I have set up CallCentric as my SIP Trunk
host=callcentric.com
username=17772048668
secret=password
type=peer
and I have created an outbound route that uses it with a dial plan that should let all calls go through.
The logs from the Voicent System show
"SIP OUTGOING:----Remote Host:10.27.73.119---- Port: 5060----\nINVITE sip:[email protected]:5060 SIP/2.0\r\nFrom: \"789\";tag=1c12069\r\nTo: sip:[email protected]:5060\r\nCall-Id: vx_0_0_4_T_9400\r\nCSeq: 12342 INVITE\r\nContact: \"789\"\r\nContent-Type: application/sdp\r\nContent-Length: 189\r\nMax-Forwards: 70\r\nUser-Agent: Voicent Soft Phone\r\nAllow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, REGISTER, SUBSCRIBE, NOTIFY\r\nSupported: replaces\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-82ad712212a6;rport\r\n\r\nv=0\r\no=sipX 5 8 IN IP4 10.27.73.116\r\ns=call\r\nc=IN IP4 10.27.73.116\r\nt=0 0\r\nm=audio 9156 RTP/AVP 0 101\r\na=rtpmap:0 pcmu/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-15\r\na=ptime:20\r\n--------------------END--------------------\n"
12:58:44.965 T-04972:
"SIP INCOMING:----Remote Host:10.27.73.119---- Port: 5060----\nSIP/2.0 401 Unauthorized\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-82ad712212a6;received=10.27.73.116;rport=5060\r\nFrom: “789"sip:[email protected]:5060;tag=1c12069\r\nTo: sip:[email protected]:5060;tag=as56fdb265\r\nCall-ID: vx_0_0_4_T_9400\r\nCSeq: 12342 INVITE\r\nServer: FPBX-2.11.0beta2(11.3.0)\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nWWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“31a83020”\r\nContent-Length: 0\r\n\r\n====================END====================\n”
12:58:44.968 T-04972:
"SIP OUTGOING:----Remote Host:10.27.73.119---- Port: 5060----\nACK sip:[email protected]:5060 SIP/2.0\r\nContact: sip:[email protected]\r\nFrom: “789"sip:[email protected]:5060;tag=1c12069\r\nTo: sip:[email protected]:5060;tag=as56fdb265\r\nCall-Id: vx_0_0_4_T_9400\r\nCSeq: 12342 ACK\r\nMax-Forwards: 70\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-82ad712212a6;rport\r\nContent-Length: 0\r\n\r\n--------------------END--------------------\n”
12:58:44.969 T-09400: Call (4) created
12:58:44.975 T-04972:
"SIP OUTGOING:----Remote Host:10.27.73.119---- Port: 5060----\nINVITE sip:[email protected]:5060 SIP/2.0\r\nFrom: "789"sip:[email protected]:5060;tag=1c12069\r\nTo: sip:[email protected]:5060\r\nCall-Id: vx_0_0_4_T_9400\r\nCSeq: 12343 INVITE\r\nContact: “789"sip:[email protected]\r\nContent-Type: application/sdp\r\nContent-Length: 189\r\nMax-Forwards: 70\r\nUser-Agent: Voicent Soft Phone\r\nAllow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, REGISTER, SUBSCRIBE, NOTIFY\r\nSupported: replaces\r\nAuthorization: Digest username=“789”, realm=“asterisk”, nonce=“31a83020”, uri=“sip:[email protected]:5060”, response=“834122854a4c2b0eae67931fc50bc80d”, algorithm=MD5\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-3ce16b6c149e;rport\r\n\r\nv=0\r\no=sipX 5 8 IN IP4 10.27.73.116\r\ns=call\r\nc=IN IP4 10.27.73.116\r\nt=0 0\r\nm=audio 9156 RTP/AVP 0 101\r\na=rtpmap:0 pcmu/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-15\r\na=ptime:20\r\n--------------------END--------------------\n”
12:58:44.977 T-04972:
"SIP INCOMING:----Remote Host:10.27.73.119---- Port: 5060----\nSIP/2.0 100 Trying\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-3ce16b6c149e;received=10.27.73.116;rport=5060\r\nFrom: “789"sip:[email protected]:5060;tag=1c12069\r\nTo: sip:[email protected]:5060\r\nCall-ID: vx_0_0_4_T_9400\r\nCSeq: 12343 INVITE\r\nServer: FPBX-2.11.0beta2(11.3.0)\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContact: sip:[email protected]:5060\r\nContent-Length: 0\r\n\r\n====================END====================\n”
12:58:44.980 T-04972:
"SIP INCOMING:----Remote Host:10.27.73.119---- Port: 5060----\nSIP/2.0 183 Session Progress\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-3ce16b6c149e;received=10.27.73.116;rport=5060\r\nFrom: “789"sip:[email protected]:5060;tag=1c12069\r\nTo: sip:[email protected]:5060;tag=as08220771\r\nCall-ID: vx_0_0_4_T_9400\r\nCSeq: 12343 INVITE\r\nServer: FPBX-2.11.0beta2(11.3.0)\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContact: sip:[email protected]:5060\r\nContent-Type: application/sdp\r\nContent-Length: 233\r\n\r\nv=0\r\no=root 923407895 923407895 IN IP4 10.27.73.119\r\ns=Asterisk PBX 11.3.0\r\nc=IN IP4 10.27.73.119\r\nt=0 0\r\nm=audio 13492 RTP/AVP 0 101\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n====================END====================\n”
12:58:52.864 T-04972:
"SIP INCOMING:----Remote Host:10.27.73.119---- Port: 5060----\nSIP/2.0 503 Service Unavailable\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-3ce16b6c149e;received=10.27.73.116;rport=5060\r\nFrom: “789"sip:[email protected]:5060;tag=1c12069\r\nTo: sip:[email protected]:5060;tag=as08220771\r\nCall-ID: vx_0_0_4_T_9400\r\nCSeq: 12343 INVITE\r\nServer: FPBX-2.11.0beta2(11.3.0)\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\r\nSupported: replaces, timer\r\nContent-Length: 0\r\n\r\n====================END====================\n”
12:58:52.867 T-04972:
"SIP OUTGOING:----Remote Host:10.27.73.119---- Port: 5060----\nACK sip:[email protected]:5060 SIP/2.0\r\nContact: sip:[email protected]\r\nFrom: “789"sip:[email protected]:5060;tag=1c12069\r\nTo: sip:[email protected]:5060;tag=as08220771\r\nCall-Id: vx_0_0_4_T_9400\r\nCSeq: 12343 ACK\r\nMax-Forwards: 70\r\nVia: SIP/2.0/UDP 10.27.73.116;branch=z9hG4bK-3ce16b6c149e;rport\r\nContent-Length: 0\r\n\r\n--------------------END--------------------\n”
Are there any logs to show call routes? I have no idea if it is bad configuration for a route, or if my CallCentric information is set up incorrectly. Voicent can use CallCentric directly, but I want more control over my calls.