Audio problem on Twinned calls

[Tech2014]

We have an IPO running 11.1 at a client that has SIP trunks; some of the users twin to their cell phone.

They notified me today that when someone calls through to the user's DID number and they answer the twinned call on their cell phone,
neither party gets audio. If someone at an internal extension calls to the same twinned extension, they get audio with the user on the
cell phone. I watched in System status and I see both call types use RTP Relay, so it wasn't a problem with Direct Media Path (which
I've had in the past once or twice), and I can even do an unconditional forward on the extension to the cell phone number, and calls
have audio both ways.

Anyone have a suggestion on what to check for this? I would have thought maybe something on their network was blocking RTP ports, but
if the calls that work and the calls that don't are both over RTP Relay, it would seem that nothing can be blocking those ports.

Thanks for any help you can give me.

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[IPOTS]

Check the SIP URI and set the Forwarding/Twinning to "Caller

 

[Nortel4Ever]

Try enabling Keepalives under LAN1 or 2 (which everone you're using) - VOIP - scroll down to Keepalives

 

[Tech2014]

Thanks to both of you. I kindof wondered about the URI setup. I will also check Keepalives.

 

[vid151]

I am seeing this on release 11.1.2.3.0 , I know I got this working before the last update... seems to be one of those reoccurring items that crops up on here. I have keep alives on, and changed the forwarding URI ( this just changes the caller ID to the twinned station rather than passthrough the caller id) did not help with audio.

 

[Tech2014]

I've had a lot of other things come up the last day, but I did look in URI and I may need to change some settings and test. I do have Keepalives enabled, with a periodic timeout of 30. This did work until we changed to a different kind of SIP trunk, although this client changes their network a lot so it could be coincidence. I am going to use the URI settings I was using for the older SIP trunks and test with them tomorrow and see if it makes a difference. If it works I will post exactly what my setup is in this thread.