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Transfer Hmp Voiceguide Dont Work


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#1 Diego Miņo 22 March 2016 - 06:53 AM

Hi.
voiceguide server 192.168.1.50
ext 200
ext 205

fortivoice (PBX ) server 192.168.1.199
ext 206
ext 203

 

 

When I call ext 200 and 205 voiceguide works.
When voiceguide call ext 206 and 203 works .

Does not work is the transfer to extensions 203 or 206 SIP

Please show your support

 

 



#2 Diego Miņo 22 March 2016 - 07:20 AM

Neither works: transferring 203@192.168.1.199

as hookflash or dialout and conference



#3 SupportTeam 22 March 2016 - 09:27 AM

The  0321_1509_vgEngine.zip  trace captures a REFER transfer attempt,  but the destination is specified is:

 

203

 

most likely the full SIP address is required. Please try using:

 

203@192.168.1.199

 

 

The Fortivoice PBX at 192.168.1.199 is accepting the REFER transfer to just "203", so maybe you should look in that PBXs logs to see why it accepted the transfer without actually later connecting it, but using the full SIP address may resolve the issue that you have.

  • ws2.png


#4 SupportTeam 22 March 2016 - 09:41 AM

The WireShark in trace 0321_1607_vgEngine - Copy.zip  captures a Trombone Transfer (Dial and Conference Transfer) attempt.

 

The outgoing leg of the '2-line tromboned' call transfer is rejected by Fortivoice PBX at 192.168.1.199 with a "407 - PROXY AUTHORIZATION REQUIRED".

 

This is because the CallerID was not set on the outgoing call leg - so the CallerID from the incoming call was re-used (206@192.168.0.199) - but as Fortivoice requests authentication before placing outgoing calls the outgoing call failed as VoiceGuide does not have authentication details for that relayed Caller ID.

 

You need to specify CallerID on outgoing calls that is one of the extensions that is registered by VoiceGuide.

This is done using the <CallerID> option in the VoiceGuide's Transfer module, or in the Config.xml file.

see: http://www.voiceguid...tml/modxfer.htm

Once the outgoing CallerID is specified then the authentication on outgoing call will be made as well, and the outgoing leg of the call will be connected.

 

 

 

 

Easiest approach to set up the system is to just configure Fortivoice to accept incoming calls from VoiceGuide's IP 192.168.1.50

 

Basically setting up SIP trunk between VoiceGuide and Fortivoice.

 

This way Fortivoice will just accept all incoming calls from VoiceGuide's IP 192.168.1.50.

 

 

 

 

 

Another approach is:

 

You can setup an authentication that VoiceGuide can use with Fortivoice when sending calls to it. Then if VoiceGuide receives a 407 response from Fortivoice then VoiceGuide will re-issue the SIP invite to Fortivoice with the right authentication to let Fortivoice accept the call.

 

If you will setup a user on Fortivoice that VoiceGuide can use, then those details can be set in VoiceGuide's Config.xml file.

Please see: http://www.voiceguid...ip_register.htm

 

Then VoiceGuide will register itself with Fortivoice and authenticate any 407 responses.

 

 

As mentioned, its just easiest to set up a  SIP trunk between VoiceGuide and Fortivoice, to have Fortivoice just accept all incoming calls from VoiceGuide's IP: 192.168.1.50

  • ws3.png


#5 Diego Miņo 23 March 2016 - 02:01 AM

Adjunto el archivo config.xml usado.

 

Están configuradas 2 extensiones.   200 y 205

 

 

<VoIP_Registrations>
<VoIP_Registration>

<Protocol>SIP</Protocol>
<RegServer>192.168.1.199</RegServer>
<RegClient>200@192.168.1.199</RegClient>
<LocalAlias>200@192.168.1.50</LocalAlias>
<CallerID>200@192.168.1.199</CallerID>
</VoIP_Registration>

<VoIP_Registration>
<Protocol>SIP</Protocol>
<RegServer>192.168.1.199</RegServer>
<RegClient>205@192.168.1.199</RegClient>
<LocalAlias>205@192.168.1.50</LocalAlias>
<CallerID>205@192.168.1.199</CallerID>
</VoIP_Registration>

 

</VoIP_Registrations>

<VoIP_Authentications>
<VoIP_Authentication>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>200</AuthUsername>
<AuthPassword>200</AuthPassword>
</VoIP_Authentication>

<VoIP_Authentication>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>205</AuthUsername>
<AuthPassword>205</AuthPassword>
</VoIP_Authentication>

</VoIP_Authentications>

 

Diego



#6 SupportTeam 23 March 2016 - 05:55 AM

Could you please post the WireShark trace that captures the VoiceGuide service startup and the transfer attempts. This will let us see if the registration/authentication setting specified are getting accepted by PBX.

 

Also please post the VoiceGuide traces that capture the service startup and the transfer attempts.



#7 Diego Miņo 01 April 2016 - 03:39 AM

Attached as requested.
Not transfer to 203@192.168.1.199

 

Br

 

Diego

  • Attached File  logs.zip   220.11KB   68 downloads


#8 SupportTeam 01 April 2016 - 07:24 AM

WireShark logs shows registration of extensions 200 and 205 was successful, and your Fortivoice screenshot shows them as registered as well, so all is good there.

 

And the REFER transfer was ACCEPTED by Fortivoice (like it was before).

 

 

Can you post a capture of a 'Dial and Conference' type transfer to 203@192.168.1.199 ? (Please include service startup time in trace - which will show us the registrations as well)

 

 

Did you ask Fortivoice as to why their system is advising that they have ACCEPTED the REFER transfer but did not complete the transfer of call? You can show them the WireShark traces attached which show that Fortivoice ACCEPTED the REFER transfer.

  • fortivoice registrations and refer trasnfer.png


#9 SupportTeam 01 April 2016 - 09:04 AM

It would be good to see a WireShark capture of a call transfer from one FortiVoice extension to another.

If you can post a capture of that then we can then see if there is any other information in the REFER packets sent from a FortiVoice telephone when the FortiVoice telephone is doing the REFER transfer.

Please ensure that all communications for all legs of call is included in that capture.

 

 

 

The supplied WireShark trace in your previously posted logs.zip does not include the communications between extension 206 (at IP 192.168.1.147) and FortiVoice (at IP 192.168.1.199). That SIP connection would have had to be established before FortiVoice sent a call INVITE to IVR (extension 200). So why isn't that captured in the WireShark log ?

 

Are you able to advise why those SIP packets were not captured? This may explain why we also do not see any packets from FortiVoice (at IP 192.168.1.199) to extension 203 (at IP 192.168.1.200) after FortiVoice accepted the REFER trasnfer...

Is extension 203 working and you are able to place calls between 203 and 206? Are you able place calls between 203 and 200/205 (the IVR) ?