frederick Posted January 16, 2009 Report Share Posted January 16, 2009 I see in the general settings the Inband DTMF detection: ON; But when I tried to do an inbound call with Inband-DTMF, the pbxnsip cannot detect this. What could be wrong? Also I did Log Level 9 for general logging and log media events ON, but can't get the logs I would like to analyze for Inband-DTMF. When I used DTMF-rfc2833 on my device, the DTMF works! and I can get some logs (example: [6] 2009/01/16 10:02:46: Received DTMF 1 ). Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 16, 2009 Report Share Posted January 16, 2009 I see in the general settings the Inband DTMF detection: ON;But when I tried to do an inbound call with Inband-DTMF, the pbxnsip cannot detect this. What could be wrong? Also I did Log Level 9 for general logging and log media events ON, but can't get the logs I would like to analyze for Inband-DTMF. When I used DTMF-rfc2833 on my device, the DTMF works! and I can get some logs (example: [6] 2009/01/16 10:02:46: Received DTMF 1 ). A difficult topic. I believe if the user agent advertizes RFC2833 (the new number is actually RFC4733) the PBX has no motivation to burn CPU resources on analyzing it. Yes, you should see "DTMF: Power:" on log level 9 (Media). Quote Link to comment Share on other sites More sharing options...
frederick Posted January 16, 2009 Author Report Share Posted January 16, 2009 A difficult topic. I believe if the user agent advertizes RFC2833 (the new number is actually RFC4733) the PBX has no motivation to burn CPU resources on analyzing it. Yes, you should see "DTMF: Power:" on log level 9 (Media). Does this mean that if a Voip-Device communicates DTMF-Inband to our PBXnSIP server, the dtmf will not be detected? Which particular settings we can look to make it work (both rfc2833 and inband)? Thanks again. Quote Link to comment Share on other sites More sharing options...
frederick Posted January 26, 2009 Author Report Share Posted January 26, 2009 I just want to brought back this issue if there's a way that PBXnSIP support InBand DTMF? If yes, what configuration we can do to make it work? Thanks again. Quote Link to comment Share on other sites More sharing options...
mabbott Posted April 21, 2009 Report Share Posted April 21, 2009 A VOIP provider I am working with requires using inband dtmf if 711 is used as the codec, but it doesn't work because they are offering RFC2833 DTMF. Is there any way around this? They suggested switching to 729, but my license doesn't have 729 enabled. What is the best solution? Quote Link to comment Share on other sites More sharing options...
pbx support Posted April 21, 2009 Report Share Posted April 21, 2009 A VOIP provider I am working with requires using inband dtmf if 711 is used as the codec, but it doesn't work because they are offering RFC2833 DTMF. Is there any way around this? They suggested switching to 729, but my license doesn't have 729 enabled. What is the best solution? I am not sure what to say here.. why do they offer RFC2833 when they ask to you use inband DTMF? pbxnsip works fine with both in-band and out-of-band(end to end). What we did not support is one leg in-band and other leg out of band. But we have added that support now and is in the testing phase. Quote Link to comment Share on other sites More sharing options...
mabbott Posted April 21, 2009 Report Share Posted April 21, 2009 That's a very good question. It's a carrier I think you guys work/worked with closely, Prescient Worldwide. Is there a way to force inband on all calls, even if 2833 is offered? Quote Link to comment Share on other sites More sharing options...
mabbott Posted April 22, 2009 Report Share Posted April 22, 2009 They gave me a little more information about what is going on. They say "when the call is re-INVITEd back to G.711 there is no RFC2833 DTMF listed in the SDP (and it should start detecting inband DTMF at that point)" Below are the logs from the call. Any help would be appreciated. INVITE sip:1877658xxxx@209.190.245.xxx:5060 SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060> i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed m: <sip:+1603870xxxx-GXcomtechtollgold-n3u3u5j4d1eo2@207.138.151.33:5060;transport=udp> k: timer x: 64800 Min-SE: 64800 l: 292 Content-Disposition: session; handling=required c: application/sdp v=0 o=Sonus_UAC 78310 7831000 IN IP4 207.138.151.38 s=SIP Media Capabilities c=IN IP4 207.138.151.38 t=0 0 m=audio 12714 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sendrecv a=maxptime:20 [9] 2009/04/22 09:47:37: UDP: Opening socket on 0.0.0.0:50876 [9] 2009/04/22 09:47:37: UDP: Opening socket on 0.0.0.0:50877 [5] 2009/04/22 09:47:37: Identify trunk (IP address/port and domain match) 4 [9] 2009/04/22 09:47:37: Resolve 142564: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142564: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142564: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Content-Length: 0 [6] 2009/04/22 09:47:37: Sending RTP for 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx#813034ca74 to 207.138.151.38:12714 [5] 2009/04/22 09:47:37: Trunk ITSP (not global) sends call to account 90 in domain localhost [7] 2009/04/22 09:47:37: Attendant: Set language to first language en [8] 2009/04/22 09:47:37: Play recordings/att11.wav space20 [9] 2009/04/22 09:47:37: Resolve 142565: udp 209.190.198.110 34766 [6] 2009/04/22 09:47:37: send codec=pcmu/8000 [9] 2009/04/22 09:47:37: Resolve 142566: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142566: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142566: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbx/3.3.1.3177 Content-Type: application/sdp Content-Length: 230 v=0 o=- 312390002 312390002 IN IP4 209.190.245.xxx s=- c=IN IP4 209.190.245.xxx t=0 0 m=audio 50876 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=sendrecv [6] 2009/04/22 09:47:37: send codec=pcmu/8000 [9] 2009/04/22 09:47:37: Resolve 142567: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142567: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142567: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbx/3.3.1.3177 Content-Type: application/sdp Content-Length: 230 v=0 o=- 312390002 312390002 IN IP4 209.190.245.xxx s=- c=IN IP4 209.190.245.xxx t=0 0 m=audio 50876 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=sendrecv [9] 2009/04/22 09:47:37: SIP Rx udp:207.138.151.33:5060: ACK sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK7e66eb355ab66842cfe2f0e95d8b02d9-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 ACK Max-Forwards: 70 l: 0 [9] 2009/04/22 09:47:37: SIP Rx udp:207.138.151.33:5060: INVITE sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK8789d802947ade0d00ef54990918c76a-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41512 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed m: <sip:+1603870xxxx-GXcomtechtollgold-n3u3u5j4d1eo2@207.138.151.33:5060;transport=udp> k: timer x: 64800;refresher=uac Min-SE: 64800 l: 186 Content-Disposition: session; handling=required c: application/sdp v=0 o=Sonus_UAC 78310 7831001 IN IP4 207.138.151.38 s=SIP Media Capabilities c=IN IP4 207.138.151.38 t=0 0 m=audio 12714 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=maxptime:20 [6] 2009/04/22 09:47:37: send codec=pcmu/8000 [9] 2009/04/22 09:47:37: Resolve 142568: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142568: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142568: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK8789d802947ade0d00ef54990918c76a-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41512 INVITE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbx/3.3.1.3177 Content-Type: application/sdp Content-Length: 150 v=0 o=- 312390002 312390002 IN IP4 209.190.245.xxx s=- c=IN IP4 209.190.245.xxx t=0 0 m=audio 50876 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=sendrecv [6] 2009/04/22 09:47:37: Call hold from trunk [9] 2009/04/22 09:47:38: SIP Rx udp:207.138.151.33:5060: ACK sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bKc271aed5a61a5ea57eddb2122c8d680b-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41512 ACK Max-Forwards: 70 l: 0 [9] 2009/04/22 09:47:39: Resolve 142569: udp 200.105.211.92 40930 [9] 2009/04/22 09:47:40: Resolve 142570: udp 209.190.198.110 35920 [9] 2009/04/22 09:47:40: Resolve 142571: udp 200.105.211.92 8254 [9] 2009/04/22 09:47:41: Resolve 142572: udp 209.190.198.110 34766 [9] 2009/04/22 09:47:41: Resolve 142573: udp 209.190.198.110 33742 [9] 2009/04/22 09:47:41: Resolve 142574: udp 209.190.198.110 33742 [9] 2009/04/22 09:47:42: SIP Rx udp:207.138.151.33:5060: BYE sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK12bf9d14dafeee9ef90293720ae76780-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41513 BYE Max-Forwards: 70 l: 0 [9] 2009/04/22 09:47:42: Resolve 142575: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:42: Resolve 142575: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:42: Resolve 142575: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:42: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK12bf9d14dafeee9ef90293720ae76780-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41513 BYE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> User-Agent: pbx/3.3.1.3177 RTP-RxStat: Dur=4,Pkt=207,Oct=35604,Underun=0 RTP-TxStat: Dur=4,Pkt=210,Oct=36120 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
mabbott Posted April 23, 2009 Report Share Posted April 23, 2009 Is there a setting for force to inband? Or would this require a new build for a fix? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 18, 2009 Report Share Posted May 18, 2009 Is there a setting for force to inband? Or would this require a new build for a fix? Seems that some switched have a problem if the OOB DTMF codec is not the same as they propose. There is a fix available in head; maybe you can get a build (ask Pradeep) and see if that fixes the problem. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 18, 2009 Report Share Posted May 18, 2009 Is there a setting for force to inband? Or would this require a new build for a fix? Seems that some switched have a problem if the OOB DTMF codec is not the same as they propose. There is a fix available in head; maybe you can get a build (ask Pradeep) and see if that fixes the problem. Quote Link to comment Share on other sites More sharing options...
pbx support Posted May 18, 2009 Report Share Posted May 18, 2009 Seems that some switched have a problem if the OOB DTMF codec is not the same as they propose. There is a fix available in head; maybe you can get a build (ask Pradeep) and see if that fixes the problem. Win32: http://pbxnsip.com/protect/pbxctrl-3.3.2.3183.exe. Please let's know if you needed build for other OS Quote Link to comment Share on other sites More sharing options...
mabbott Posted May 18, 2009 Report Share Posted May 18, 2009 I would need a build for a CS410. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 18, 2009 Report Share Posted May 18, 2009 I would need a build for a CS410. http://pbxnsip.com/cs410/update-4.0.0.3204.tgz (beta!!!) Quote Link to comment Share on other sites More sharing options...
mabbott Posted May 26, 2009 Report Share Posted May 26, 2009 Can I get this in a 3.3 cs410 version to test? The carrier won't give me a number to test the beta with and I can't test this with the customers equipment. Quote Link to comment Share on other sites More sharing options...
pbx support Posted May 26, 2009 Report Share Posted May 26, 2009 Can I get this in a 3.3 cs410 version to test? The carrier won't give me a number to test the beta with and I can't test this with the customers equipment. Please try http://pbxnsip.com/cs410/update-3.4.0.3191.tgz Quote Link to comment Share on other sites More sharing options...
mabbott Posted May 29, 2009 Report Share Posted May 29, 2009 The 3.4 build did not solve the problem. It still isn't switching back to inband dtmf when 2833 is removed from the re-invite. Quote Link to comment Share on other sites More sharing options...
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