Frank.Schwarz Report post Posted 05/07/2024 05:21 AM Hello Voicegudie Support Team, I am trying a test SIP installation. I have installed HMP on my computer and with microSIP I can connect to the SIP line. I have attached the Confi.xml and a Whireshark recording. Can someone tell me what is wrong? Thank you very much! Frank 2024-05-06_Wireshark.pcapng Config.xml Share this post Link to post
SupportTeam Report post Posted 05/07/2024 07:16 AM WireShark trace shows that you are trying to perfrom a SIP REGISTER process with this PBX at 192.168.20.179: Swyx IpPbxSrv/13.31 (Swyx.Core_13.31_20240219.1) but the Swyx IpPbx immediately replies with "401 Unauthorized" even after Dialogic HMP sends the Authorization Digest, and keeps on doing so: Pleases see this previous thread on Registering with Swyx IpPbx : https://www.voiceguide.com/forums/topic/14013-register-vg-sip/?tab=comments#comment-45907 see the second post in that thread for instructions on how Swyx IpPbx account had to setup to allow successful SIP Registers from Dialogic HMP. Share this post Link to post
Frank.Schwarz Report post Posted 05/17/2024 10:53 AM Hello Voiceguide Team, It seems that the line is now registered. However, there is no call on the Voiceguide. Please find attached the loggings (Wireshark/SWYX/VoiceGuide). Can you please check this? Thanks a lot Frank 2024-05-17_Logging.zip Share this post Link to post
SupportTeam Report post Posted 05/17/2024 10:06 PM Supplied WireShark trace shows that Swyx PBX is replying with an OK to a SIP REGISTER sent to it from 192.168.20.75 The WireShark trace is not showing any calls (ie. any SIP INVITEs) being sent from Swyx to IP 192.168.20.75 You should contact the Swyx PBX supplier and ask them why that PBX is not sending the calls. Share this post Link to post
Frank.Schwarz Report post Posted 05/29/2024 07:46 AM Hello VG support team, the line is now registered and the call is being forwarded. Apparently the test license has now expired. Is there a way to extend it? Please see the attachment. Thank you!!! 2024-05-29 090735.zip Share this post Link to post
SupportTeam Report post Posted 05/29/2024 08:53 AM Your VoiceGuide installation is right now in "Free Evaluation" mode, as you have just downloaded and installed VoiceGuide, but have not yet purchased a license for your system. In "Free Evaluation" mode you need to periodically restart the VoiceGuide Service to keep it working, and number of calls (incoming and outgoing) is limited. The "Free Evaluation" lets you fully test the system before purchasing licenses. You will need to restart the VoiceGuide service to have VoiceGuide working for another ~20 minutes (depending on version). You can right click on the "IVR" icon in the icon tray at the bottom right of the screen to Stop and then Start the service, or you can also use the Windows' Services applet to do this. Share this post Link to post
Frank.Schwarz Report post Posted 06/05/2024 06:32 AM I restarted the PC and then restarted the VoiceGuide service. No communication with Swyx in WireShark. All loggings are attached. What is going wrong here? Thank you!!! 2024-06-05.zip Share this post Link to post
Frank.Schwarz Report post Posted 06/05/2024 06:42 AM Sorry for the last post, I can't reach SWYX at the moment Share this post Link to post
SupportTeam Report post Posted 06/05/2024 06:45 AM The ktTel trace shows this error report from Dialogic drivers: 195 082551.643 7552 WARN IPEC_REG_FAIL_internalError : internal IP CCLib error encountered while attempting to formulate an outgoing REGISTER request. Maybe Dialogic's RTF log is showing more information on why HMP has returned that error. Sounds like you currently have a lack of network connectivity between HMP and your PBX. That would explain it. Share this post Link to post
Frank.Schwarz Report post Posted 06/13/2024 06:25 AM Hello support team, I now have a connection to SWYX but my test script is not playing. I restarted the VG service before the test. I have attached the loggings. Can you check what the problem could be. Thank you very much! 2024-06-12 Logging.zip Share this post Link to post
SupportTeam Report post Posted 06/13/2024 11:44 AM The "Start Module" of the script that you are using is set to the module named "Auflegen". The module named "Auflegen" is a 'Hangup' type module... This is why VoiceGuide hangs up the call immediately after answering it... In the Script Deisigner go to "Edit"->"Script Properties" to set the "Start Module" : From looking at the vgEngine trace you can also see that the first module ran after answering the calls is "Auflegen" : 155217.220 24 3 1 2 ev CallState GCEV_ANSWERED, crn=8000001, iEvent=0 ,256,1,4, s1:, s2:, s3:, build_date: 2022-06-18 18:34:35.66 155217.220 24 3 1 2 q_scr + evCallState lcode=0 scode=[GCEV_ANSWERED] 155217.220 19 3 1 2 q_scr run evScriptEvent 2050 GCEV_ANSWERED action_id=0, crn=8000001 [2050|0|0|0|0][|||||] 00:00:00 max:1|0,5079 155217.220 19 3 1 2 evscr GCEV_ANSWERED 2050 2050|0|0 || LineState=LS_OFFERED_ANSWERINGCALL 155217.220 19 3 1 2 q_scr run evCallState 0 GCEV_ANSWERED action_id=0, crn=8000001 [256|1|4|0|0][|||||] 00:00:00 max:1|0,5079 155217.220 19 3 1 2 callstate crn_event=8000001 0|GCEV_ANSWERED state=256|Connected calldirection=1 gcCallState=4 [||], ScriptState=LS_OFFERED_ANSWERINGCALL 155217.220 19 3 1 2 LineEvCallState L1_current=Offered crn_event=8000001,ev=0,GCEV_ANSWERED,1, sParam1= 155217.220 19 3 1 2 LineEvCallState CONNECTED begin 155217.220 19 3 1 2 set crn_connected=8000001 visual_ivr_session_id=842364669 (called from LINECALLSTATE_CONNECTED) 155217.220 19 3 1 2 L1_set Connected (iScriptState=LS_OFFERED_ANSWERINGCALL) 155217.220 19 3 1 2 LineEvCallState_Connected_InBound begin 155217.220 19 3 1 2 Inband detection not enabled 155217.221 19 3 1 2 L2_set Running_Normal (LineEvCallState_Connected_InBound_ScriptStartsNow) 155217.221 19 3 1 2 StartLoadedVgs 1 : C:\INSOCAM\Test.vgs (7.6.43 - 7.6.8204.33438 2022-06-18 18:34:35.66) 155217.221 19 3 1 2 set sScriptToRunOnHangup=[] in StartLoadedVgs 155217.221 19 3 1 2 rv add $RV_STARTTIME|2024-06-12 15:52:17 155217.221 19 3 1 2 rv add $RV_DEVICEID|3 155217.221 19 3 1 2 rv add DlgcVoice|dxxxB1C1 155217.221 19 3 1 2 rv add DlgcNetwork|iptB1T1 155217.221 19 3 1 2 rv add $RV_CIDNAME| 155217.221 19 3 1 2 rv add DNIS|700@192.168.20.75 155217.221 19 3 1 2 rv add ISDN_SETUP|[sip_header_request_uri]{sip:700@192.168.20.75}[sip_header_contact_uri]{sip:006813836170@192.168.20.179:5060;transport=udp}[sip_header_from_display]{"INSOCAM AES Leitstellensoftware"}[sip_header_to_display]{"700"}[sip_header_callid]{R8VYDx-Fak2A-afoeXMZAQ..}[sip_header_from]{"INSOCAM AES Leitstellensoftware"<sip:006813836170@hws.local;user=phone>;tag=8515ed2b}[sip_header_to]{"700"<sip:700@192.168.20.75>}[sip_header_Via]{SIP/2.0/UDP 192.168.20.179:5060;rport=5060;branch=z9hG4bK-524287-1---064de0679d61ed68}[sip_header_From]{"INSOCAM AES Leitstellensoftware"<sip:006813836170@hws.local;user=phone>;tag=8515ed2b}[sip_header_To]{"700"<sip:700@192.168.20.75>}[sip_header_Contact]{<sip:006813836170@192.168.20.179:5060;transport=udp>}[sip_header_Call-ID]{R8VYDx-Fak2A-afoeXMZAQ..}[sip_header_User-Agent]{Swyx IpPbxSrv/13.31 (Swyx.Core_13.31_20240219.1)} 155217.221 19 3 1 2 rvns add PathApp|C:\Program Files (x86)\VoiceGuide 155217.221 19 3 1 2 rvns add PathVoiceGuide|C:\Program Files (x86)\VoiceGuide 155217.221 19 3 1 2 rvns add ScriptPath|C:\INSOCAM\ 155217.221 19 3 1 2 rvns add ScriptsPath|C:\INSOCAM\ 155217.221 19 3 1 2 rvns add PathScript|C:\INSOCAM 155217.221 19 3 1 2 rv add $RV_CIDNUMBER|006813836170@hws.local 155217.222 19 3 1 2 t timer clear force=False(RunModule_begin) 155217.222 19 3 1 2 RunModule start Hangup the Call, [Auflegen], vgm=1, previous_vgm=1 Share this post Link to post
Frank.Schwarz Report post Posted 06/14/2024 07:27 AM Hello support team, that was the mistake! Thank you!!! Share this post Link to post
Frank.Schwarz Report post Posted 07/09/2024 02:34 PM Hello VoiceGuide Team, I can now establish a connection to the telephone system with the VG. The call is answered and a wave file is also played. However, no DTMF tones arrive at the VG. They can be seen in WireShark. Can you have a look? Attached is the logging. Thank you very much 2024-07-09_VG_Log.zip Share this post Link to post
SupportTeam Report post Posted 07/10/2024 08:46 AM Looks like this PBX (Swyx IpPbxSrv/13.31) right now transmits the DTMF events as 'INFO' messages. Please set it to use RFC2833 to send DTMF events. Also, please update system to latest version of VoiceGuide. We can only correctly interpret traces when the latest version of VoiceGuide is used. Share this post Link to post
Frank.Schwarz Report post Posted 08/06/2024 01:29 PM Hello VoiceGuide Team, I have the following feedback from my customer: =============================== Hello Mr. Schwarz, We have received feedback from Swyx: For internal devices, DTMF is only transmitted via SIP info. Can this be activated in Voiceguide? Alternatively, I have just tried to register the SIP trunk directly in the IVR. I have the same issue as when registering with Swyx. The authentication is not transmitted correctly. Please find the Wireshark attached. Can you ask the manufacturer about this again? After the first login, the second one with the authentication header is missing. First login Voiceguide: See Picture 1 Nothing happens after that. If the trunk is registered on the Swyx, a second one with the Auth header is sent after the first Login See Picture 2 Also attached is the Wireshark log ============================== Thanks!!! Yours sincerely Frank 2024-08-06.zip Share this post Link to post
SupportTeam Report post Posted 08/13/2024 05:02 AM RFC2883 is the standard way of sending DTMF signals. All the various ways in which SIP level INFO messages are used by some systems to signal DTMF events are considered as proprietary and have not been standardized. Use of SIP INFO for DTMF is not currently supported. Share this post Link to post
SupportTeam Report post Posted 08/26/2024 11:26 PM Support for SIP INFO will be added to future versions of the publicly released VoiceGuide software. Until that time can contact support@voiceguide.com directly to obtain VoiceGuide versions that support SIP INFO DTMF signaling that meet your requirements. Share this post Link to post