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hi do you have any experience with Genesys Sip Server?

Configuring a sip trunk in Genesys I send the call to voiceguide without problem, but to return or transfer I can only by the conference option. But there I occupy two ports.
 Is there a way to transfer it to a sip trunk? Or can you create a sip trunk on the voiceguide side?




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Does the SIP server that you are using supports accepting SIP REFER Transfers (RFC 3515)  to transfer its call to another URI ?

To perform the RFC3515 REFER transfer select "Blind Hookflash Transfer" option in the 'Transfer Call' module, and specify the IP/URI address of the transfer destination.

Can you use WireShark to capture the SIP communications of the call arriving to VoiceGuide from Genesys, VoiceGuide issuing a SIP REFER request to Genesys, and the Genesys' response?

Please post the WireShark .pcapng file here and we can then see what is happening.

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The WireShark trace was not attached. Can you please try attaching it again.

Please also attach VoiceGuide's vgEngine and ktTel traces that capture service startup and the transfers.

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The WireShark seems to show 2 separate calls.

Call 1 arrives from - which is an Audiocodes gateway, not Genesys. VoiceGuide issues a REFER request back to Audiocodes a IP address Audiocodes accepts the REFER transfer, but looks like that transfer fails as Audiocodes cannot find URI 989193162@, with NOTIFY messages ultimately advising '404 not found'.  You should look at Audiocodes' setup to see why that Audiocodes gateway was not able to connect to "989193162" at it's own IP address. At no stage does VoiceGuide communicate with a Genesys system, only with Audiocodes gateway.

Call 2 again arrives from Audiocodes gateway - but this time VoiceGuide then make another call on another channel out to 31010@ - which is a Genesys system. Genesys answers the call, and looks like VoiceGuide then bridges the two calls together, with the original caller hanging up 25 seconds later. From your description it look like this call was successfully connected with both parties hearing each other, correct? Genesys replies with a 481 when VoiceGuide hangs up the Genesys leg of the call, which is a bit puzzling, (maybe the extension 31010 was already hung up, but Genesys did not befoehand send any SIP message on that event?)



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dminof wrote:

hello thank you very much for your answer and help.

yes, now I understand, the call comes through audiocodes to voice guide.  Voiceguide should call back the audiocodes so that audiocodes will send it to genesys.  Is there any necessary configuration in the audiocodes to achieve this on the audiocodes trunk? Or should it be handled with the sip extension registered in voiceguide for the transfer?

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Is there any necessary configuration in the audiocodes to achieve this on the audiocodes trunk?

You should contact Audiocodes support regarding this.

It looks like for now you have a working solution with the "Dial and Converence" bridged type transfer, so you can continue to use that in the interim until you are able to get Audiocodes to handle the REFER transfer requests as required.

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