CarlH Posted December 2, 2008 Report Share Posted December 2, 2008 Hi, We are having problems with some international numbers that we have registered to forward calls to our helpdesk. E.g here are extracts from the log when calling to the Norwegian nr +4721031332 and to the Switzerland nr +41435000151. Both these nr are forwarded to 0842014000 which is the Swedish nr. When calling from the Norweigan nr i get connected but I don't hear anything. When I call to the Switz nr i hear the IVR loud and clear. Any help would be greatly appreciated! Here is a an extract from the log when calling to the Norwegian nr. [7] 2008/12/02 09:24:03:SIP Rx udp:195.149.148.40:5060: BYE sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0 Record-Route: <sip:195.149.148.40;ftag=as7d8c22a2;lr=on> Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bKa34d.10e4dbe6.0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK35364621;rport=5060 From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as7d8c22a2 To: <sip:4721031332@x.rtcfactory.com>;tag=f5150fa6b6 Call-ID: 11b0347b27f2a7164edf7e932101625f@195.138.212.41 CSeq: 103 BYE User-Agent: RTC Gateway 2.0 Max-Forwards: 70 Content-Length: 0 P-hint: call from pstn gateway [9] 2008/12/02 09:24:03:Resolve 29794699: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:24:03:Resolve 29794699: a udp 195.149.148.40 5060 [9] 2008/12/02 09:24:03:Resolve 29794699: udp 195.149.148.40 5060 [7] 2008/12/02 09:24:03:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bKa34d.10e4dbe6.0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK35364621;rport=5060 Record-Route: <sip:195.149.148.40;ftag=as7d8c22a2;lr=on> From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as7d8c22a2 To: <sip:4721031332@x.rtcfactory.com>;tag=f5150fa6b6 Call-ID: 11b0347b27f2a7164edf7e932101625f@195.138.212.41 CSeq: 103 BYE Contact: <sip:0842014000@83.145.6.141:5060;transport=udp> User-Agent: pbxnsip-PBX/3.0.1.3023 RTP-RxStat: Dur=9,Pkt=430,Oct=73960,Underun=0 RTP-TxStat: Dur=9,Pkt=441,Oct=75852 Content-Length: 0 Here is a an extract from the log when calling to the Switz nr. [7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: INVITE sip:0842014000@83.145.6.141:5060;transport=udp;line=02e74f10 SIP/2.0 Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on> Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060 From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486 To: <sip:41435000151@x.rtcfactory.com> Contact: <sip:0046707960416@195.138.212.41> Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41 CSeq: 102 INVITE User-Agent: RTC Gateway 2.0 Max-Forwards: 70 Date: Tue, 02 Dec 2008 08:10:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 268 P-hint: call from pstn gateway P-hint: local sip call v=0 o=root 14929 14929 IN IP4 195.138.212.41 s=session c=IN IP4 195.138.212.41 t=0 0 m=audio 15002 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [9] 2008/12/02 09:10:59:UDP: Opening socket on port 50050 [9] 2008/12/02 09:10:59:UDP: Opening socket on port 50051 [5] 2008/12/02 09:10:59:Identify trunk (line match) 27 [9] 2008/12/02 09:10:59:Resolve 29788667: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788667: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788667: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060 Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on> From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486 To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470 Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41 CSeq: 102 INVITE Content-Length: 0 [7] 2008/12/02 09:10:59:Set packet length to 20 [6] 2008/12/02 09:10:59:Sending RTP for 033369853e67a81277f6e86855db9b4f@195.138.212.41#670aff6470 to 195.138.212.41:15002 [5] 2008/12/02 09:10:59:Trunk RTC 0842014000 sends call to 00 in domain smarthost.se [8] 2008/12/02 09:10:59:Play recordings/ivr79.wav [7] 2008/12/02 09:10:59:Set packet length to 20 [9] 2008/12/02 09:10:59:Resolve 29788668: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788668: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788668: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060 Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on> From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486 To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470 Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41 CSeq: 102 INVITE Contact: <sip:0842014000@83.145.6.141:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Type: application/sdp Content-Length: 228 v=0 o=- 46299 46299 IN IP4 83.145.6.141 s=- c=IN IP4 83.145.6.141 t=0 0 m=audio 50050 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [9] 2008/12/02 09:10:59:Resolve 29788669: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788669: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788669: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060 Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on> From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486 To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470 Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41 CSeq: 102 INVITE Contact: <sip:0842014000@83.145.6.141:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Type: application/sdp Content-Length: 228 v=0 o=- 46299 46299 IN IP4 83.145.6.141 s=- c=IN IP4 83.145.6.141 t=0 0 m=audio 50050 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: ACK sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0 Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on> Via: SIP/2.0/UDP 195.149.148.40;branch=0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK5faf7a1a;rport=5060 From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486 To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470 Contact: <sip:0046707960416@195.138.212.41> Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41 CSeq: 102 ACK User-Agent: RTC Gateway 2.0 Max-Forwards: 70 Content-Length: 0 P-hint: call from pstn gateway [7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: ACK sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0 Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on> Via: SIP/2.0/UDP 195.149.148.40;branch=0 Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK3cda2bc1;rport=5060 From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486 To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470 Contact: <sip:0046707960416@195.138.212.41> Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41 CSeq: 102 ACK User-Agent: RTC Gateway 2.0 Max-Forwards: 70 Content-Length: 0 P-hint: call from pstn gateway Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 2, 2008 Report Share Posted December 2, 2008 Yes, that whole topic is addressed in 3.1.1. If you can, get a 3-minute demo key, set up a test server and try the latest & greatest (http://pbxnsip.com/protect/pbxctrl-3.1.1.3100.exe). Then set your contry code to 47 (if I am right here) and then the numbers should be formatting correctly - automatically. Quote Link to comment Share on other sites More sharing options...
CarlH Posted January 12, 2009 Author Report Share Posted January 12, 2009 Yes, that whole topic is addressed in 3.1.1. If you can, get a 3-minute demo key, set up a test server and try the latest & greatest (http://pbxnsip.com/protect/pbxctrl-3.1.1.3100.exe). Then set your contry code to 47 (if I am right here) and then the numbers should be formatting correctly - automatically. Hi! I have now upgraded to the latest version but we still have the same problem with numbers forwarded from Norway and UK. Any thoughts? All help is very appreciated. Br Carl Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 12, 2009 Report Share Posted January 12, 2009 I have now upgraded to the latest version but we still have the same problem with numbers forwarded from Norway and UK. Any thoughts? All help is very appreciated. Looking at the old messages I must admit I don't exactly get what the problem is. Is it a problem related to RTP or is it a problem related to the phone number (+47 or 0047 or 01147 and so on)? Maybe you can get a fresh LOG... Quote Link to comment Share on other sites More sharing options...
CarlH Posted January 12, 2009 Author Report Share Posted January 12, 2009 Looking at the old messages I must admit I don't exactly get what the problem is. Is it a problem related to RTP or is it a problem related to the phone number (+47 or 0047 or 01147 and so on)? Maybe you can get a fresh LOG... Well, Ill try to explain more in detail. Our trunk provider has registered numbers in different countries in europe which are forwarded to our support line. All numbers except those for UK, Norway and Finland work fine. The provider has tried forwarding directly to one of their phones and it works. When those numbers are forwarded to our support line (an IVR) all we here is silence. The call is connected and I can see it in "calls" but we can't hear the IVR. Denmark works Finland doesn't. DENMARK (45) NATIONAL 1 +4569918175 FINLAND (358) HELSINKI (9) 1 +358942419025 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 12, 2009 Report Share Posted January 12, 2009 Well, Ill try to explain more in detail. Our trunk provider has registered numbers in different countries in europe which are forwarded to our support line. All numbers except those for UK, Norway and Finland work fine. The provider has tried forwarding directly to one of their phones and it works. When those numbers are forwarded to our support line (an IVR) all we here is silence. The call is connected and I can see it in "calls" but we can't hear the IVR. Denmark works Finland doesn't. DENMARK (45) NATIONAL 1 +4569918175 FINLAND (358) HELSINKI (9) 1 +358942419025 Try forcing a specific codec on the trunk. Probably the provider has a problem when the PBX answeres with more than one codec (see http://wiki.pbxnsip.com/index.php/One-way_Audio). Quote Link to comment Share on other sites More sharing options...
CarlH Posted January 12, 2009 Author Report Share Posted January 12, 2009 Try forcing a specific codec on the trunk. Probably the provider has a problem when the PBX answeres with more than one codec (see http://wiki.pbxnsip.com/index.php/One-way_Audio). Which codecs do you recommend? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 12, 2009 Report Share Posted January 12, 2009 Which codecs do you recommend? I would just take 0 (ulaw). Quote Link to comment Share on other sites More sharing options...
CarlH Posted January 12, 2009 Author Report Share Posted January 12, 2009 I would just take 0 (ulaw). It works now . Thanks! Quote Link to comment Share on other sites More sharing options...
Recommended Posts
Join the conversation
You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.