shopcomputer Posted February 8, 2008 Report Share Posted February 8, 2008 I am trying to figure out what I am doing wrong, I am using the tel:alias for the first time for a DID block provided by the carrier. I set alias names to tel:3474244022 for extension 22, it is not working, and this is what I see in the logs. If I set the trunk to send all calls to the auto-attendands extension then this new DID block does work. [0] 2008/02/08 09:37:28: SIP Rx udp:64.152.60.75:5060: INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222> Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:9174170964@64.152.60.75:5060> P-Asserted-Identity: <sip:9174170964@64.152.60.75:5060> Supported: timer Session-Expires: 1800 Min-SE: 90 Content-Length: 279 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 3495 31360 IN IP4 64.152.60.75 s=SIP Media Capabilities c=IN IP4 64.152.60.164 t=0 0 m=audio 8280 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 [7] 2008/02/08 09:37:28: UDP: Opening socket on port 50448 [7] 2008/02/08 09:37:28: UDP: Opening socket on port 50449 [0] 2008/02/08 09:37:28: SIP Tx udp:64.152.60.75:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 INVITE Content-Length: 0 [7] 2008/02/08 09:37:28: Set packet length to 20 [6] 2008/02/08 09:37:28: Sending RTP for 1712050395_129292298@64.152.60.75#adad1c9d5d to 64.152.60.164:8280 [5] 2008/02/08 09:37:28: Received incoming call without trunk information and user has not been found [7] 2008/02/08 09:37:28: Set packet length to 20 [0] 2008/02/08 09:37:28: SIP Tx udp:64.152.60.75:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 INVITE Contact: <sip:3474244022@64.113.246.222:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [0] 2008/02/08 09:37:28: Last message repeated 2 times [0] 2008/02/08 09:37:28: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 ACK Max-Forwards: 70 Content-Length: 0 [7] 2008/02/08 09:37:28: Other Ports: 2 [7] 2008/02/08 09:37:28: Call Port: 3c47348884cd-5rqcm28316wh#24f2d0d27e [7] 2008/02/08 09:37:28: Call Port: 3ed31811@pbx#28804 [0] 2008/02/08 09:37:28: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 ACK Max-Forwards: 70 Content-Length: 0 [0] 2008/02/08 09:37:29: SIP Rx udp:64.152.60.75:5060: INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222> Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:9174170964@64.152.60.75:5060> P-Asserted-Identity: <sip:9174170964@64.152.60.75:5060> Supported: timer Session-Expires: 1800 Min-SE: 90 Content-Length: 279 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 15387 5494 IN IP4 64.152.60.75 s=SIP Media Capabilities c=IN IP4 64.152.60.71 t=0 0 m=audio 12958 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 [7] 2008/02/08 09:37:29: UDP: Opening socket on port 53930 [7] 2008/02/08 09:37:29: UDP: Opening socket on port 53931 [0] 2008/02/08 09:37:29: SIP Tx udp:64.152.60.75:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 INVITE Content-Length: 0 [7] 2008/02/08 09:37:29: Set packet length to 20 [6] 2008/02/08 09:37:29: Sending RTP for 1712050397_64827812@64.152.60.75#cb5aae4b1a to 64.152.60.71:12958 [5] 2008/02/08 09:37:29: Received incoming call without trunk information and user has not been found [7] 2008/02/08 09:37:29: Set packet length to 20 [0] 2008/02/08 09:37:29: SIP Tx udp:64.152.60.75:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 INVITE Contact: <sip:3474244022@64.113.246.222:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [0] 2008/02/08 09:37:29: Last message repeated 2 times [0] 2008/02/08 09:37:29: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 ACK Max-Forwards: 70 Content-Length: 0 [7] 2008/02/08 09:37:29: Other Ports: 2 [7] 2008/02/08 09:37:29: Call Port: 3c47348884cd-5rqcm28316wh#24f2d0d27e [7] 2008/02/08 09:37:29: Call Port: 3ed31811@pbx#28804 [0] 2008/02/08 09:37:29: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 ACK Max-Forwards: 70 Content-Length: 0 [0] 2008/02/08 09:37:30: SIP Rx udp:64.152.60.75:5060: INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222> Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:9174170964@64.152.60.75:5060> P-Asserted-Identity: <sip:9174170964@64.152.60.75:5060> Supported: timer Session-Expires: 1800 Min-SE: 90 Content-Length: 279 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 9420 9690 IN IP4 64.152.60.75 s=SIP Media Capabilities c=IN IP4 64.152.60.164 t=0 0 m=audio 23172 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 [7] 2008/02/08 09:37:30: UDP: Opening socket on port 55196 [7] 2008/02/08 09:37:30: UDP: Opening socket on port 55197 [0] 2008/02/08 09:37:30: SIP Tx udp:64.152.60.75:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 INVITE Content-Length: 0 [7] 2008/02/08 09:37:30: Set packet length to 20 [6] 2008/02/08 09:37:30: Sending RTP for 1712050399_42141136@64.152.60.75#050f7bfefc to 64.152.60.164:23172 [5] 2008/02/08 09:37:30: Received incoming call without trunk information and user has not been found [7] 2008/02/08 09:37:30: Set packet length to 20 [0] 2008/02/08 09:37:30: SIP Tx udp:64.152.60.75:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 INVITE Contact: <sip:3474244022@64.113.246.222:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [0] 2008/02/08 09:37:30: Last message repeated 2 times [0] 2008/02/08 09:37:30: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 ACK Max-Forwards: 70 Content-Length: 0 [7] 2008/02/08 09:37:30: Other Ports: 2 [7] 2008/02/08 09:37:30: Call Port: 3c47348884cd-5rqcm28316wh#24f2d0d27e [7] 2008/02/08 09:37:30: Call Port: 3ed31811@pbx#28804 [0] 2008/02/08 09:37:30: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 ACK Max-Forwards: 70 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 8, 2008 Report Share Posted February 8, 2008 [5] 2008/02/08 09:37:28: Received incoming call without trunk information and user has not been found That seems to be the problem. Are you using an outbound proxy on the trunk? This outbound trunk must also resolve to the "inbound" trunk - the PBX needs to IP address of the trunk to determine that the calls is supposed to come from that trunk. Quote Link to comment Share on other sites More sharing options...
shopcomputer Posted February 8, 2008 Author Report Share Posted February 8, 2008 Outbound Proxy on the trunk is set to east.ga.broadvox.net which is successfully resolving to 64.152.60.75 the from address on this call. Quote Link to comment Share on other sites More sharing options...
shopcomputer Posted February 8, 2008 Author Report Share Posted February 8, 2008 Just an update, if I change the outbound proxy to the IP instead of east.ga.broadvox.net it works, should it not query DNS, I do receive that IP when I ping it. I rather use DNS name as it does change as the have redundant systems. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 9, 2008 Report Share Posted February 9, 2008 Sometimes providers send from different addresses than they receive from. This is not very NAT friendly and it makes it very difficult to authenticate incoming requests. In this case it makes sense to have one trunk for outbound and another one for inbould. Quote Link to comment Share on other sites More sharing options...
jag Posted February 13, 2008 Report Share Posted February 13, 2008 Try setting a Domain alias of the IP address of the trunk.. INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Create an domain ALIAS of 64.113.246.222 for the domain, that may solve the problem. Quote Link to comment Share on other sites More sharing options...
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