IPO Call fwd to cell - No audio

[j.flanagan]

I have a customer with an IP500v2 on 7.0(36) sip trunks.
Initially, when unconditional forward was turned on, external calls would ring at the users phone and then just get dumped into their vm if not answered.
After digging around on here a little while I tried a few things out and finally was able to get the calls to forward by populating the "originator number for forwarded and twinned calls" box with their main number.
So, i can call in with my desk or cell phone and the call gets forwarded back out to my lab phone but when I pick up the call there is no audio.
I checked the RTP keep alive and that is already enabled.
What else do I need to be looking at?

Thanks

 

[derfloh]

How is network Topology configured?

IP Office remote service
Fixed price SIP trunk configuration
CLI based call blocking
SCN fallback over PSTN

 

[j.flanagan]

There's absolutely nothing configured under the network topo tabs of either lan1 or lan2.

 

[derfloh]

Probably that's the issue. It should reflect your firewall settings and your public IP.

IP Office remote service
Fixed price SIP trunk configuration
CLI based call blocking
SCN fallback over PSTN

 

[j.flanagan]

I'll get in touch with there network team and try to collect the information I need for the network topo. Normal calls are working fine both in and out. It's only the forwarded calls that are connecting with no audio.

 

[j.flanagan]

Alright, so, their sip trunks are off an SBC at their data center, therefore no STUN required. I pulled some traces which I'll attach to this thread. I used a few different filter sets, I wasn't exactly sure what was needed.

 

[j.flanagan]

First Trace

 

[j.flanagan]

Second Trace

 

[j.flanagan]

Third Trace

 

[janni78]

There's no description on what the traces are, basically you just need trace with SIP Tx/Rx enabled.

But I wouldn't run SIP trunks on any release before R9, and preferable even newer releases.
A lot has changed with SIP trunking in the last years.

"Trying is the first step to failure..." - Homer

 

[Ekster]

Go to the LAN settings for the interface the SIP trunk is using and in the VOIP tab set:


"Keep Alives - Scope" to RTP
"Initial Keep Alives" to Enabled
"Periodic Timeout" to 10 seconds.

Might be worth making sure "Re-Invite Supported" is enabled on the SIP trunk VOIP Tab too.

“Some humans would do anything to see if it was possible to do it.
If you put a large switch in some cave somewhere, with a sign on it saying 'End-of-the-World Switch. PLEASE DO NOT TOUCH'.
The paint wouldn't even have time to dry.”

Terry Pratchet