Release Line + Dialogic 1000 Media Gateway

[lucasklos]

Hello Support Team, 

We have been able to Run Voiceguide 7.4, in an Environment with HMP, receiving the calls from a Dialogic Media Gateway.

It works Ok, but if calling from analog phones, and hang up the call before the script go to the hang up module, the script keep running locking the line.

Please find attached some logs and wireshark traces taken from the computer running voiceguide + hmp.

Here you will find a call, that after a few voice messages, caller pressed 1 2 3 and then hang up. But this hangup event was not recognized by voiceguide/HMP.

Could you please take a look?

Thanks

0204_1344_vgEngine.txt

0204_CallEvents.txt

0204_ktTel.txt

0204_ktTts.txt

ivrserver_wireshark.pcapng

[SupportTeam]

VoiceGuide will hangup if it receives a signal that the other side of the call hung up.

Looks like in this case no explicit SIP Disconnect message was sent to VoiceGuide. You would need to look at the rest of your telephony network to establish whether you can set it up in a way where it does send a Disconnect message upon remote side hangup.

Do you know if perhaps a 'disconnected' tone is played on the line after remote side hangup? To check if tone is played just use Record type module to record what is happening on the line and hangup remote side when script is in that module. If a tone is played then you can set up VoiceGuide to detect that tone. Please see:

https://www.voiceguide.com/vghelp/source/html/disconnectiondetect.htm

 

If there is no disconnect indication of any kind available then the script just needs to designed in way that it times out after a certain time of no input from caller.

Supplied vgEngine trace shows that module [EnterUserCode] asked caller for input 6 times, playing (SoundFiles\201_PersonalCode.wav) 6 times before line was hung up from Line Status Monitor:

134456.312   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)
134513.102   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)
134520.936   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)
134529.514   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)
134537.438   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)
134546.018   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)
134553.941   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)
134553.979  25   3   1 wcf   IPortStatus Action_Hangup call. iPort=1

If there is no disconnect indication of any kind available then this [EnterUserCode] module (and any other similar) should have their number of replays limited to smaller number if you want line to be hung up sooner

On the calls supplied, it looks like module [EnterUserCode] was on it's last replay:

134549.514   6   3   1       iCurrReplayCount=4, iMaxReplayCount=5
...
134553.941   6   3   1 state [EnterUserCode] Playing wav (SoundFiles\201_PersonalCode.wav)

So that call would have been hung up by VoiceGuide about 10 seconds later.

 

Here is WireShark trace that shows no SIP Disconnect (BYE) was sent to VoiceGuide:

[lucasklos]

Thanks.

I extracted the RTP stream from the Wireshark trace and used audacity to analyze the disconnect tone. Added it to configline.xml and after restarting voiceguide the disconnect tone was recognized. 

[SupportTeam]

OK, thanks for letting us know how you resolved this issue.

[Guest mohammedzoheb]

Dear Lucasklos,

I would like to know what exactly you in Configline.xml that would be helpful for me.

[lucasklos]

Hello mohammedzoheb,

I followed the guide for detecting disconect tone (from the link provided by support team in a previous post) and added the following Tone in my ConfigLine.xml
 

<Tone Name="DISCONNECT USER 1">
<Notes>Disconnect Tone</Notes>
<ID>DISCONNECT_USER_1</ID>
<Freq1>424</Freq1>
<Freq1Dev>40</Freq1Dev>
<Freq2>0</Freq2>
<Freq2Dev>0</Freq2Dev>
<On>50</On>
<OnDev>5</OnDev>
<Off>50</Off>
<OffDev>5</OffDev>
<Count>3</Count>
</Tone>

This of course can be quite different depending on the pbx and the architecture you are using.

Regards