3300 to 5000 SIP Intercom just static

[jeremywatco]

Hi all,

working out some kinks.. starting to link all our remote 5000 sites up to our main 3300 site via SIP. When placing an intercom call from the 3300 to an extension on the 5000 it rings just fine on the other side however when someone picks up the 3300 side just gets really loud static and the 5000 side gets just dead air. You cannot hear anything if anyone talks either.

Any ideas?

Thanks!

[jeremywatco]

In doing more testing I have a thought...

Are SIP phone calls treated differently than Intercom calls? Example.. when forwarding a phone call from the outside over our telco sip trucks to a remote 5000 location there arent any issues. But from my desk if I dial the extension I want at the remote 5000 location I get just garbage...

Possibly the compression settings are different? Probably since the call coming in is already compressed from the Telco? Is there a way to set the compression of the intercom calls vs the phone calls?

Strange thing is intercom calls from 5000 back to 3300 are perfect.


Thanks

[LoopyLou]

You might be on to something there regarding codec differences. Any chance you can do a wireshark trace to see whats going on. I know that MCD 6 added a new codec. Maybe the 5000 software doesn't support it yet.

[jeremywatco]

OK here is an update.. my mind is boggled!

If I am sitting on the 3300 side and dial an extension on the 5000 side I get static.

If I am on the 5000 side, the extensions arent setup, so If I dial the 3300 SIP trunk number and then dial the extension the call goes through perfectly.

Now... to complicate (or maybe simplify) it further.

 If I setup a speed call from one of our DID's on the 3300 to ring through to the 5000 its rings through with caller ID on the 5000. When the 5000 answers it, I can hear the person on the 5000 side but they hear nothing.

On the 5000 side if I make a call across the SIP trunk by dialing the SIp trunk number (just like my intercom call above) again I can here the person on the 5000 side however they cannot hear me.

Any thoughts? Strange thing is I have a site with the exact same setup that works perfectly.

[marcolive]

Try to attach a Wireshark trace so we can see what happen under the hood! 

[jeremywatco]

After thinking about this overnight... I really am leaning more towards possibly a route needing to be added inside our Telco's SIP gateway. Thats the only thing that is making sense to me right now as data is free flowing between the systems... if voice traffic enters the Gateway and it doesnt know how to get to our remote node then it will just drop it.. or in this case give one way audio.... at least that sounds good to me.

[jeremywatco]

Update * Issue resolved for outgoing & incomming SIP calls. TELCO needed to add routes to edge router for remote sites.

Thanks for all the help