Genesys CTI User Forum

Hi Cavagnaro
hope all is well.
thank you for replying.
I am using GAX v9.0.109.04 (latest). Smaller agent exports work, just not larger ones (ie >500 agents).
It is also reported that users cannot make more than 1 change to a PG at a time, so they have to close the object then reopen and make the second or subsequent change. I cant see any related option but this was not an issue in our previous 850 version.

thanks

I have a GAX query, on agent export limit.
wondered if anyone experienced this issue on v9.0
thanks

Hi Victor. We might have found something in trunks by Alcatel side that we added (the SDP reuse thing in Genesys).

I set the "reuse-sdp-on-reinvite" option in all Alcatel trunks to force the SDP in the re-invite to the Alcatel.

Result:

When the reuse-sdp-on-reinvite option is set on the Alcatel trunk and I generate a wait call in queue, during the RequestRouteCall, the SIP Server re-queries the endpoint to retrieve the SDP and sends the invite to the Alcatel on the ERP with the SDP.
Result: the agents can be heard in both sides.

When I remove the option from the trunk and generate a wait, during the RequestRouteCall, the SIP Server sends the invite directly to the Alcatel on the ERP without the SDP.
Result: the SIP agent can hear but not the called Alcatel agent.

I will keep update here, thank you.

[quote author=Kubig link=topic=12632.msg55813#msg55813 date=1744810986]
Direct RTP is enabled or disabled on Alcatel level? Have you tried to change this settings?
[/quote]

Hi, yes direct RTP must be enabled on Alcatel side, we cannot disabled it because it is the main enterprise switch already in place.

[quote author=Kubig link=topic=12632.msg55810#msg55810 date=1744803855]
Do you see the incoming RTP on both sides? Are both call-legs with same codec? Are the RTP destinations correct?
[/quote]

Hi Kubig how are you ?
Well there is obviously something strange somewhere. The RTP are correct on both side yes, both legs have not same codec (Genesys Softphone is PCMA 8000 in priority, and Alcatel is G722), but they are listed on the list and should be correctly negociated on SDP). What is weird is the RTP destinations, when the audio is not audible, the RTP flow is direct from SIP agent to Alcatel agent, but in the other side, the RTP flow from Alcatel agent seems to go on an Alcatel card until the Alcatel agent holds the call to force the RTP flow to go to the SIP agent.
We tried everything on Alcatel side, and also tried to change the ISCC route type to direct no token, but it did not work.