Test Script On Voiceguide/hmp

[Guest ThaoHuynh]

Hi VG team,

I have installed Dialogic HMP and VoiceGuide Sysem again. It's worked for me. When calling VoiceGuide using SJPhone, nothing happens on the script side as if i pressed Client ID and pressed hash key in final (like: 123#). In SJPhone, i'm using rfc 2833.

I attached wireshark and kTel trace files in the same call.

Please check attached files.

 

Thao Huynh

WireShark+ log.zip

[SupportTeam]

WireShark shows that no RTP is being received from 192.168.76.37

 

Most likely the firewall on your system is blocking it.

 

Please open UDP and TCP ports 49152 to 49200.

 

You should see RTP packets from 192.168.76.37 in WireShark trace interleaved with RTP from 192.168.76.21 (which is there now)

[Guest ThaoHuynh]

I have turned off firewall system.

How can open UDP and TCP ports 49152 to 49200?

Why voiceguide hang up automatically very fast? I not yet pressed key.

[SupportTeam]

Can you now see RTP from 192.168.76.37 ?

 

If firewall is turned off there is no need to open the ports in the firewall for now.

 

You can press keys any time during the sound file playback as well, there is no need wait till end of message.

 

 

 

The default Credit Card Payment script is set to hangup 5 seconds after end of play.

 

This can be changed by editing the script: C:\Program Files (x86)\VoiceGuide\Scripts\Credit Card Payment\Credit Card Payment.vgs

 

Here is vgEngine trace extract showing system timing out after 5 sec after end of play:

154410.465   9   3   1 state <offered> : cid=192.168.76.37 , dnis=192.168.76.21 connected
154410.518   9   3   1 state [Get Client ID] Playing wav (PayGetId.wav)
154419.022  22   3   1 ev    PlayEnd 3 605750 [bytes_played=0:0, bytes_in_file=66785]
154424.495   4   3   1 timer EV_TIMEOUT_ENTERDATA
154424.507   9   3   1 state Hanging up... [No data entered]

[Guest ThaoHuynh]

No, i can't see RTP packets from 192.168.76.37 to 192.168.76.21.

It just from 192.168.76.21 to 192.168.76.37, not otherwise.

[SupportTeam]

You will need to establish why the RTP is not being sent to the VoiceGuide system from your softphone.

 

Perhaps try placing calls from different softphones on different systems.

 

Might be worth checking if there are any anti-virus type applications or other VoIP applications on either system that could be blocking the RTP traffic from arriving at VoiceGuide system.

 

 

Monitoring SIP/RTP traffic (using WireShark) on softphone machine will let you see if the RTP is sent out from the softphone.

 

Monitoring SIP/RTP traffic (using WireShark) on network from other devices attached to same network will let you see if the RTP is sent on the ethernet or not.

 

This should let you narrow down where the 192.168.76.37 -> 192.168.76.21 direction RTP traffic is being stopped/blocked

[SupportTeam]

Is it possible that ESXi's own firewall is blocking the 192.168.76.37 -> 192.168.76.21 direction RTP traffic ?

[Guest ThaoHuynh]

I use 2 physical machine, i also using WireShark to capture RTP packets in SJPhone machine and other devices (i run voiceguide system in this machine).

It still block the 192.168.76.37 -> 192.168.76.21 direction RTP traffic.

I have unstalled anti-virus type application, and no VoIP applications.

[SupportTeam]

So Windows is installed directly on the machine hardware and there is no ESXi involved on either machine in this setup?

 

Was the WireShark trace that was posted in the first post of this thread captured on the SJPhone machine or on the VoiceGuide machine?

 

Suggest running WireShark on both machines at same time until this RTP issue is resolved.

 

If for some reason you think it might be SJPhone that is not doing RTP transmission then you could try using Linphone instead: http://www.linphone.org/

[Guest ThaoHuynh]

 

 

So Windows is installed directly on the machine hardware and there is no ESXi involved on either machine in this setup?

Yes.

 

 

Was the WireShark trace that was posted in the first post of this thread captured on the SJPhone machine or on the VoiceGuide machine?

It's on VoiceGuide machine.

 

 

If for some reason you think it might be SJPhone that is not doing RTP transmission then you could try using Linphone instead: http://www.linphone.org/

If i want to use Linphone instead, i need to config my XML like http://www.voiceguide.com/vghelp/source/html/config_voip_register.htm?

[SupportTeam]

Suggest running WireShark on both machines at same time until this RTP issue is resolved. This will give you a better picture of what is going on.

 

Linphone can dial IP address direct, just like SJPhone.

 

Three is no need to set up any SIP registrations on VoiceGuide or Linphone. Just use Linphone to directly dial the IP address of VoiceGuide system.

[Guest ThaoHuynh]

I dial IP address 192.168.76.21 it will display 1921687621. And when i dial sip:192.168.76.21, it display 192.168.76.21.

It not working. WireShark can't capture any SIP or RTP packets in Linphone system and VoiceGuide system

 

[SupportTeam]

You should be seeing SIP messages from Linphone.

 

if you are not seeing any SIP message from Linphone then perhaps something else on that Linphone system is interfering with the Linphone operation or the IP packet sending.

 

Do you perhaps still have SJPhone installed on that system? You should only have one SIP application installed on system. If you have multiple different applications installed then the chances are they will be interfering with each other.

 

Was anything else installed on that Linphone machine apart from Windows+WireShark+Linphone?

 

Was anything else installed on the SJPhone machine apart from Windows+WireShark+SJPhone?

 

 

Please wee screenshot of Linphone we just installed on a system here which is placing an outgoing direct IP call. Nothing is answering that call, but you can see that SIP messages are captured in WireShark.

 

 

Sounds like you had better success with SJPhone, perhaps its a better idea to go back to SJPhone (uninstall Linphone first) and try to work out what could be blocking the RTP traffic from reaching VoiceGuide.

 

Suggest running WireShark on both machines at same time when placing a call into system to better see where the RTP traffic is blocked.

[SupportTeam]

If you are having trouble placing an outgoing 'Direct IP' call from your system(s) then do you perhaps have access to a physical SIP phone? Most physical VoIP/SIP phones will let you dial IP address as well.

 

eg: see these instructions: http://www.grandstream.com/support/faq/common-questions/direct-ip-call

[Guest ThaoHuynh]

I did it. I have unstalled sjphone 1.65 and installed another versions. So WireShark can capture RTP packet from 192.168.76.37

Thank you so much.

[SupportTeam]

Thanks for letting us know the calls are working now.

 

Can you advise which other version did you install?

[Guest ThaoHuynh]

I installed sjphone 1.6 instead.