Hubertus R. Posted October 3, 2012 Report Share Posted October 3, 2012 Hello, I'm having troubles in getting a Bintec R1200 to run as a gateway for incoming calls. I successfully set up the connection using it as a SIP Registration. Also the Call Identification from P-Asserted-Identify works fine. Only the routing to the correct extension doesn't work. So far I can only set a fixed routing using the "Send call to extension" field. Maybe someone can give me an advise, what I have to change to get this running. Here are the SIP messages, that might help. The called ISDN-MSN was the 25, which should be routed to the internal extension 25. [5] 2012/10/03 14:33:12: SIP Rx udp:192.168.151.250:5060: INVITE sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250> Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Max-Forwards: 70 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER Supported: 100rel, replaces, timer User-Agent: fec sip stack Allow-Events: refer, message-summary, dialog Content-Type: application/sdp Content-Length: 183 v=0 o=- 23506 1 IN IP4 192.168.151.250 s=SIP call c=IN IP4 192.168.151.250 t=0 0 m=audio 10010 RTP/AVP 8 18 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv [5] 2012/10/03 14:33:12: SIP Tx udp:192.168.151.250:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250>;tag=7a36527553 Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 INVITE Content-Length: 0 As can be seen, the system cannot find the target extension. [5] 2012/10/03 14:33:12: SIP Tx udp:192.168.151.250:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250>;tag=7a36527553 Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.150.16:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [5] 2012/10/03 14:33:12: SIP Rx udp:192.168.151.250:5060: ACK sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bKF6C5E3B3293926108DB5190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=00E51951AE3026108D91190009190038 To: "25" <sip:25@192.168.151.250>;tag=7a36527553 Call-ID: 229AE3B3293926108DB4190009190038 CSeq: 1 ACK Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Content-Length: 0 In case I set the "Send call to extension" field manually to 25, the successful invite to sip extension 25 looks like this: [5] 2012/10/03 14:30:42: SIP Tx tcp:192.168.222.26:58226: INVITE sip:25@192.168.101.135:58226;transport=tcp;line=ylef8f1x SIP/2.0 Via: SIP/2.0/TCP 192.168.150.16:5060;branch=z9hG4bK-3ffa59e0b2a97dfcc3306e4b81299598;rport From: "0123123456" <sip:0123123456@snom.mydomain;user=phone>;tag=49668 To: "Sip User 25" <sip:25@snom.mydomain> Call-ID: abdee4b8@pbx CSeq: 3565 INVITE Max-Forwards: 70 Contact: <sip:25@192.168.150.16:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 256 v=0 o=- 50613 50613 IN IP4 192.168.150.16 s=- c=IN IP4 192.168.150.16 t=0 0 m=audio 51504 RTP/AVP 8 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/10/03 14:30:42: SIP Rx tcp:192.168.222.26:58226: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.150.16:5060;branch=z9hG4bK-3ffa59e0b2a97dfcc3306e4b81299598;rport=5060 From: "0123123456" <sip:0123123456@snom.mydomain;user=phone>;tag=49668 To: "Sip User 25" <sip:25@snom.mydomain>;tag=zeczbc5izk Call-ID: abdee4b8@pbx CSeq: 3565 INVITE Contact: <sip:25@192.168.101.135:58226;transport=tcp;line=ylef8f1x>;reg-id=1 Content-Length: 0 So as long as I just want to have one MSN being sent to one extension, this is ok. But I have of course multiple ones I would like to use, without configuring a trunk for each. Best HR Quote Link to comment Share on other sites More sharing options...
Vodia support Posted October 3, 2012 Report Share Posted October 3, 2012 Can you send us another trace, please follow these guidelines. http://wiki.snomone.com/index.php?title=How_to_Take_SIP_log_file. Quote Link to comment Share on other sites More sharing options...
Hubertus R. Posted October 3, 2012 Author Report Share Posted October 3, 2012 Thanks for the quick response, her the new log: [5] 2012/10/03 16:29:07: SIP Rx udp:192.168.151.250:5060: INVITE sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250> Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Max-Forwards: 70 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER Supported: 100rel, replaces, timer User-Agent: fec sip stack Allow-Events: refer, message-summary, dialog Content-Type: application/sdp Content-Length: 182 v=0 o=- 7872 1 IN IP4 192.168.151.250 s=SIP call c=IN IP4 192.168.151.250 t=0 0 m=audio 10024 RTP/AVP 8 18 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv [8] 2012/10/03 16:29:07: Allocating for call port 15, SIP call id EEB910E5393926108DC8190009190038 [5] 2012/10/03 16:29:07: SIP Tx udp:192.168.151.250:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2 Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 INVITE Content-Length: 0 [7] 2012/10/03 16:29:07: Set packet length to 20 [6] 2012/10/03 16:29:07: Call-leg 15: Sending RTP for EEB910E5393926108DC8190009190038 to 192.168.151.250:10024, codec not set yet [8] 2012/10/03 16:29:07: Incoming call: Request URI sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c, To is "25" <sip:25@192.168.151.250> [8] 2012/10/03 16:29:07: Trunk R1200-10@snom.mydomain has country code 49, area code 2581 [8] 2012/10/03 16:29:07: call port 15: state code from 0 to 404 [7] 2012/10/03 16:29:07: Set packet length to 20 [5] 2012/10/03 16:29:07: SIP Tx udp:192.168.151.250:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport=5060 From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2 Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 INVITE Contact: <sip:snomone@192.168.150.16:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Content-Length: 0 [5] 2012/10/03 16:29:07: SIP Rx udp:192.168.151.250:5060: ACK sip:snomone@192.168.150.16:5060;transport=udp;line=1679091c SIP/2.0 Via: SIP/2.0/UDP 192.168.151.250:5060;branch=z9hG4bK46E410E5393926108DC9190009190038;rport From: "0123123456" <sip:0123123456@192.168.151.250;user=phone>;tag=5A6D7BC1373926108DBE190009190038 To: "25" <sip:25@192.168.151.250>;tag=1fade82bb2 Call-ID: EEB910E5393926108DC8190009190038 CSeq: 1 ACK Contact: <sip:snomone@192.168.151.250:5060;transport=udp> Content-Length: 0 [8] 2012/10/03 16:29:07: Clearing call port 15, SIP call id EEB910E5393926108DC8190009190038 [8] 2012/10/03 16:29:07: Hangup: Call 15 not found I replaced my phone number by 0123123456. Everything else is untouched. I already tried to figure out using the manual, but so far I'm not lucky. Our other gateways didn't make any of those problems (berofix). Thanks in advance Hubertus Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 3, 2012 Report Share Posted October 3, 2012 The problem is probably that the Request-URI contains not the destination number as with BeroFix. But that can be easily fixed on the trunk with the settings "Send call to extension". There are some examples on http://wiki.snomone.com/index.php?title=Inbounds_Calls. Quote Link to comment Share on other sites More sharing options...
Hubertus R. Posted October 4, 2012 Author Report Share Posted October 4, 2012 Thanks a lot, this solved my problem!! Quote Link to comment Share on other sites More sharing options...
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