Harry C Report post Posted 12/01/2023 02:41 AM Recently I have DTMF issue with NEC SV9500. I did have ticket with Dialogic and Dialogic support replied. The first comment from looking the below is that the ReINVITE is sent directly after the ACK received and causing an issue as well for processing in that case. Please see following technote where application needs change to handle the 491 pending situation. PowerMedia HMP rejects a re-INVITE with 491 Request Pending (dialogic.com) This call the DTMF couldn’t send from 10.92.8.13 to 10.92.38.27. DTMF.zip Share this post Link to post
SupportTeam Report post Posted 12/01/2023 03:02 AM Can you please post VoiceGuide's ktTel trace that include in it the time from 11/30/2023 13:50 till 11/30/2023 14:05. Also please post the latest ktTel trace that captures the VoiceGuide service start/restart on that system. We can then confirm current configuration and behavior on this system and advise. Please .ZIP up all trace files before posting. Share this post Link to post
SupportTeam Report post Posted 12/01/2023 03:20 AM Can you also please post VoiceGuide's vgEngine trace from Nov 29th that includes the VoiceGuide service restart. Please .ZIP up all trace files before posting. Share this post Link to post
SupportTeam Report post Posted 12/01/2023 03:35 AM Please post the other vgEngine trace from Nov 29th. The trace posted above does not include the VoiceGuide service startup, it only includes the shutdown. A new vgEngine trace file is created at time of VoiceGuide service startup. Share this post Link to post
SupportTeam Report post Posted 12/01/2023 11:50 AM We have emailed a download link to a version of VoiceGuide that now implements the "491 request pending" fix. Share this post Link to post
Harry C Report post Posted 12/06/2023 10:10 PM We still have issue with 3rd INVITE. DTMF_Issue_12062023.zip Share this post Link to post
SupportTeam Report post Posted 12/07/2023 12:34 AM In the attached DTMF_Issue_12062023.zip the ktTel trace is from 5th December (14:47 till 23:15) and the RTFLOG trace is from 6th December (00:42 till 00:57) Please post the ktTel trace from 6th December. We can then correlate that VoiceGuide ktTel trace with the posted RTFlog. Share this post Link to post
SupportTeam Report post Posted 12/07/2023 11:37 AM That highlighted "3rd INVITE" does not contain any SDP information. So there is nothing in that 3rd INVITE that indicates how DTMFs will be sent on this call, so nothing that can be 'confirmed as accepted' with respect to that either in the returned OK. The first INVITE does not contain any SDP either: The second INVITE contains an SDP, but that SDP does not include any advice on how DTMFs are sent: In cases where DTMF sending mode is not advised by the calling party it is assumed that the calling party will be sending DTMF's in-band. The "OK" reply from VoiceGuide to the 1st INVITE reflects that. (See screenshot below) You could try asking the PBX administrator if it is their intention to not send any details on how DTMFs are delivered, and how are they actually sending the DTMFs (if at all) on calls such as these - where no DTMF sending method is explicitly advised by PBX during call setup. Share this post Link to post
Harry C Report post Posted 12/07/2023 12:53 PM (edited) The 1st INVOTE doesn't contain any SDP information, but 200 OK (5465) was different with the 3rd INVITE 200 OK (5492). Why 3rd INVITE 200 OK "Media Description" doesn't have Media Format: DynamicRTP-Type-101? THIRD INVITE's OK response: FIRST INVITE's OK response: Edited 12/08/2023 05:48 AM by SupportTeam added headings to screenshots Share this post Link to post
SupportTeam Report post Posted 12/08/2023 05:35 AM Quote Why 3rd INVITE 200 OK "Media Description" doesn't have Media Format: DynamicRTP-Type-101? The ACK (5469) sent in response to first INVITEs(5464) OK(5465) contained an SDP that set 96 for the telephone-event: Media Attribute (a): rtpmap:96 telephone-event/8000 Media Attribute (a): fmtp:96 0-11 so after VoiceGuide/HMP sent "101 telephone-event" in the OK(5465), the PBX responded with a "96 telephone-event". I guess that means that even though PBX was advised that HMP is setting to use "101 telephone-event" , the PBX decided that it will be sending DTMFs using "96 telephone-event" approach. So if you need for the OK that the sent in response to the 3rd INVITE to also contain a "telephone-event" media attribute, then should we now send "96 telephone-event" or the "101 telephone-event" value? Note that sending the "101 telephone-event" again might mean going against the PBX's just previously provided "telephone-event" setting... Did the Administrator of this PBX advise what they need to see from HMP/VoiceGuide and when? Do they really need for the remote end to echo back to the PBX the value for the "telephone-event" that PBX has advised that it will be using on its connection? Did the Administrator of this PBX specifically advise that they need a "telephone-event" in that 3rd INVITE response? And if yes then what is the value they wopuld need? The one advised by PBX in its ACK, or having HMP send its original default setting (101) be OK and PBX will adjust the way it sends it's DTMFs accordingly? What about the 2nd INVITE response? Does the PBX require us to include a "telephone-event" attribute line in that OK response as well? Did the PBX Administrator specifically advise that it is lack of their PBX not seeing its "telephone-event" selection echoed back to it on the 3rd INVITE that is causing their PBX to somehow not send DTMFs? FIRST INVITE (NO SDP) 5464 ----> Session Initiation Protocol (INVITE) Request-Line: INVITE sip:22702@10.92.38.27:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK1210454877 From: "GEN INFO " <sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060> Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 1 INVITE Contact: <sip:23484@10.92.39.22> Max-Forwards: 70 User-Agent: Enterprise IP-PBX (InSIPH) Allow: INVITE, ACK, REGISTER, BYE, OPTIONS, INFO, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Date: Thu, 07 Dec 2023 01:56:14 GMT Supported: timer Session-Expires: 180;refresher=uac P-Asserted-Identity: "GEN INFO " <sip:23484@10.92.39.22> Content-Length: 0 5465 <----- Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header From: "GEN INFO "<sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 1 INVITE Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK1210454877 Supported: replaces User-Agent: MST/IVR Contact: <sip:22702@10.92.38.27:5060> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY Content-Type: application/sdp Content-Length: 223 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): Dialogic_SIP_CCLIB 273387488 273387489 IN IP4 10.92.38.27 Session Name (s): Dialogic_SIP_CCLIB Session Information (i): session information Connection Information (c): IN IP4 10.92.38.27 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 49154 RTP/AVP 0 8 101 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 [Generated Call-ID: 330d522c@10.92.39.22] 5469 -----> Session Initiation Protocol (ACK) Request-Line: ACK sip:22702@10.92.38.27:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK1893664471 From: "GEN INFO " <sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 1 ACK Contact: <sip:23484@10.92.39.22> Content-Type: application/sdp Max-Forwards: 70 User-Agent: Enterprise IP-PBX (InSIPH) Content-Length: 194 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 23484 0 1 IN IP4 10.92.39.22 Session Name (s): Std-SIP Call Connection Information (c): IN IP4 10.92.39.22 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 60304 RTP/AVP 0 96 Media Attribute (a): ptime:20 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:96 telephone-event/8000 Media Attribute (a): fmtp:96 0-11 [Generated Call-ID: 330d522c@10.92.39.22] SECOND INVITE 5470 -----> Session Initiation Protocol (INVITE) Request-Line: INVITE sip:22702@10.92.38.27:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK1031855743 From: "GEN INFO " <sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 2 INVITE Contact: <sip:23484@10.92.39.22> Content-Type: application/sdp Max-Forwards: 70 User-Agent: Enterprise IP-PBX (InSIPH) Allow: INVITE, ACK, REGISTER, BYE, OPTIONS, INFO, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Supported: timer P-Asserted-Identity: "WIRELESS CALLER" <sip:12409973549@10.92.39.22> Content-Length: 153 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 23484 0 2 IN IP4 10.92.39.22 Session Name (s): Std-SIP Call Connection Information (c): IN IP4 10.92.39.22 Time Description, active time (t): 0 0 Session Attribute (a): inactive Media Description, name and address (m): audio 60304 RTP/AVP 0 Media Attribute (a): ptime:20 Media Attribute (a): rtpmap:0 PCMU/8000 [Generated Call-ID: 330d522c@10.92.39.22] 5471 <----- Session Initiation Protocol (100) Status-Line: SIP/2.0 100 Trying Message Header From: "GEN INFO "<sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 2 INVITE Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK1031855743 Supported: replaces User-Agent: MST/IVR Contact: <sip:22702@10.92.38.27:5060> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY Content-Length: 0 5478 <----- Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header From: "GEN INFO "<sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 2 INVITE Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK1031855743 Supported: replaces User-Agent: MST/IVR Contact: <sip:22702@10.92.38.27:5060> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY Content-Type: application/sdp Content-Length: 188 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): Dialogic_SIP_CCLIB 273387488 273387490 IN IP4 10.92.38.27 Session Name (s): Dialogic_SIP_CCLIB Connection Information (c): IN IP4 10.92.38.27 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 49154 RTP/AVP 0 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): sendrecv Media Attribute (a): ptime:20 [Generated Call-ID: 330d522c@10.92.39.22] 5479 -----> Session Initiation Protocol (ACK) Request-Line: ACK sip:22702@10.92.38.27:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK0028047973 From: "GEN INFO " <sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 2 ACK Contact: <sip:23484@10.92.39.22> Max-Forwards: 70 User-Agent: Enterprise IP-PBX (InSIPH) Content-Length: 0 THIRD INVITE (NO SDP) 5489 -----> Session Initiation Protocol (INVITE) Request-Line: INVITE sip:22702@10.92.38.27:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK0020117955 From: "GEN INFO " <sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 3 INVITE Contact: <sip:23484@10.92.39.22> Max-Forwards: 70 User-Agent: Enterprise IP-PBX (InSIPH) Allow: INVITE, ACK, REGISTER, BYE, OPTIONS, INFO, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Supported: timer P-Asserted-Identity: "WIRELESS CALLER" <sip:12409973549@10.92.39.22> Content-Length: 0 5490 <----- Session Initiation Protocol (100) Status-Line: SIP/2.0 100 Trying Message Header From: "GEN INFO "<sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 3 INVITE Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK0020117955 Supported: replaces User-Agent: MST/IVR Contact: <sip:22702@10.92.38.27:5060> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY Content-Length: 0 5492 <----- Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header From: "GEN INFO "<sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 3 INVITE Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK0020117955 Supported: replaces User-Agent: MST/IVR Contact: <sip:22702@10.92.38.27:5060> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY Content-Type: application/sdp Content-Length: 144 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): Dialogic_SIP_CCLIB 273387488 273387490 IN IP4 10.92.38.27 Session Name (s): Dialogic_SIP_CCLIB Connection Information (c): IN IP4 10.92.38.27 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 49154 RTP/AVP 0 8 [Generated Call-ID: 330d522c@10.92.39.22] 5495-----> Session Initiation Protocol (ACK) Request-Line: ACK sip:22702@10.92.38.27:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.92.39.22:5060;branch=z9hG4bK2010659656 From: "GEN INFO " <sip:23484@10.92.39.22>;tag=4b223407 To: <sip:22702@10.92.39.22:5060>;tag=1e868f7d-787794c2-138f57-10483c08-0-13c4-764 Call-ID: 117b5091@10.92.39.22 [Generated Call-ID: 117b5091@10.92.39.22] CSeq: 3 ACK Contact: <sip:23484@10.92.39.22> Content-Type: application/sdp Max-Forwards: 70 User-Agent: Enterprise IP-PBX (InSIPH) Content-Length: 140 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 23484 0 3 IN IP4 10.92.39.22 Session Name (s): Std-SIP Call Connection Information (c): IN IP4 10.92.8.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 51000 RTP/AVP 0 Media Attribute (a): ptime:20 Media Attribute (a): rtpmap:0 PCMU/8000 [Generated Call-ID: 330d522c@10.92.39.22] Share this post Link to post
Harry C Report post Posted 12/08/2023 06:17 AM Dialogic did also point out that PBX send payload type 96 for rfc2833 in their ACK after we send 101 from HMP side. We did ask NEC support engineer to fix PBX side. NEC is checking on their side. How about Dialogic comment about "pass a parmblk with the IP_CAPABILITY that has the expected rfc28933 payload" Share this post Link to post
SupportTeam Report post Posted 12/08/2023 11:24 AM (edited) Can you please test Re-INVITE handling with this version: [removed] and then post the traces as before (vgEngine, ktTel and WireShark) We can then see how this version handled this case and see if further changes are needed. To change to a different version of VoiceGuide: Stop VoiceGuide Service. VoiceGuide can be stopped by clicking on the VoiceGuide Service Monitor in the Windows's Icon Tray on bottom right of the taskbar and selecting "Stop". Exit all VoiceGuide programs. This includes the Service Monitor applet in the Icon Tray area in bottom right of the screen, as well as the Script Designer, Line Status Monitor, etc. Do NOT uninstall the previous VoiceGuide installation. Stop the Dialogic service using the Dialogic Configuration Manager (DCM), or Windows' Services Applet. Run the VoiceGuide install and install into same directory as existing installation. Start VoiceGuide service. Note: Running a VoiceGuide install over the top of an existing install will NOT overwrite existing configuration or license files (Config.xml, ConfigLine.xml, VG.INI, etc) and will not remove any of users script or sound files, and will not remove any log files etc. Edited 12/13/2023 11:33 AM by SupportTeam newer version supplied to customer Share this post Link to post
Harry C Report post Posted 12/09/2023 07:17 AM the last 3 calls had the same issue. the 3rd INVITE 200 OK looks different. 1209_0135_vgEngine.zip Share this post Link to post
Harry C Report post Posted 12/10/2023 05:35 AM The attached file is RFT log file. rtflog-LOCAL-20231209-01h35m32.962s.zip Share this post Link to post
SupportTeam Report post Posted 12/11/2023 11:13 PM An email has been sent to this customer detailing next steps that need to be made on both the Dialogic side and the NEC side. Share this post Link to post
Harry C Report post Posted 01/27/2024 05:56 AM Traces files from that latest “VG.INI setup” version. ktTel and WireShark and RTF. 01272024 DTMF.zip Share this post Link to post
Harry C Report post Posted 01/29/2024 05:55 AM New Release does fix NEC SV9500 REINVITE issue. Share this post Link to post
