Incoming sip call forwarded through multisite VPN

[gdamron]

I am trying to add an extention from a remote UC560 to a blast group to the local UC540. The remote ext (82108) rings, and can be answered but no voice traffic is heard on ether side of the call. If the call originates from a FXO port and not a SIP trunk everything works fine. I would imagine it is a problem with the IP traffic from the internet address of the sip privder traversing the VPN tunnel the to remote site, but im not sure how to cure the problem. I have heard the term hairpin VPN but i don't know if that applies.

I don't know if i explained this very well but attached is a diagram that might help

[gdamron]

An update but I'm still no closer:

If a call coming from the outside sip trunk is answered at the local site first, and then forwarded to the remote site the audio works on both ends of the call. The problem is still when the outside sip call is answered first by an extension at the remote site, no audio is heard on ether end.

I would typically try packet captures to see where the data is failing but because the traffic between the local and remote site is in the VPN I don't know how to do this.

Troubleshooting Tips, Links, Suggestions would be much appreciated.

[bkwon · Accepted solution]

Hi,

basically, in the multisite with UC500, we recommend SIP trunk in each indivisual site becuase of some interowkring issues

SIP to H323 interworking not support transcoding,

- please check the codec in multisite, and SIP provider.

- if they are match then

voice class h323 1

call start interwork

[gdamron]

changing the call start on voice class h323 1 worked!

It looks like the our provider only supports g711 and g729. We are currently using g711

should I try to change the codec on the multisite dial peers or leave the h323 setting in place?

[bkwon · Accepted solution]

if the sip trunk use g711 , then multisite have to set g711 as well.

you can verify the codec info in the call legs when you make a call, and make sure they are match

"show call active  voice brief"

[gdamron]

Using the setting:

voice class h323 1

call start interwork

ended up causing problems with the SPA525G using the SSL VPN but I was able to get the codec on the multisitechanged. I this point there seems to be no problems.

Thanks for you help