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[nick at flhsi.com]

So I've got a bit of a head scratcher. Wanted to get some insight.

I've got a PBX running 12.3.0

We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office).

Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no incoming audio.

Example 1. From my T46G, I call the IP650. The call comes up in G722 and we can both talk, No issues.

Example 2. From My T46G I call outbound to my cell phone. Our SIP trunk is only G711, So The leg between the T46G and Asterisk is G722, And it has to transcode it to G711. I get no receive audio on the T46G, However the outbound audio is fine (I can hear my voice on the Cell phones receiver)

Example 3, I call inbound from my cell phone. Dial My extension. And answer. Once again, The SIP Trunk leg is G711, And the T46G leg is G722, I get audio both ways no problem.

Example 4. Exactly like Example 2, But I turn canreinvite= to yes. (I normally use no, As I wish to always remain in the audio path). Same call as Example 2 comes up, Quickly reinvites into G711 and audio works fine both ways.

And finally, I do all of the above tests from the IP650, And everything works fine. So it's limited to the T46G in some case.

My SIP config is below. Remember, I've toggled canreinvite=yes on and off. But that's just a bandaid.

I also tried this on another T46G in the office running a much older firmware and had the same issue...

[nick] type=friend context=flhsi-internal secret=REDACTED ;insecure=yes language=en canreinvite=yes host=dynamic mailbox=106@flhsi notransfer=yes dtmfmode=rfc2833 disallow=all allow=g722 allow=ulaw callerid="FLHSI Nick" <321-205-1100> nat=no call-limit=100 limitonpeer=yes


Nick Olsen Network Operations (855) FLSPEED x106