I have an Asterisk server on our LAN that serves our office VOIP
phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are
ulaw/alaw

We use attended transfer extensively. If our trunk is ulaw/alaw they work fine.

If the trunk is ilbc we have problems
1- incoming PSTN call routed via voipfone SIP down the trunk to our server
2- our phones ring ok, caller can be answered (e.g. by A)
3- A requests attended transfer to another phone (B) on the LAN-
incoming caller put on hold, A can talk to B, B can talk to A
4- A hangs up, B is connected to caller. B can hear caller, but caller
cannot hear B. Console output:
Asked to transmit frame type 64, while native formats is 0x400
(ilbc)(1024) read/write = 0x40 (slin)(64)/0x400 (ilbc)(1024)

Running Asterisk 1.6.2.9 on Ubuntu Karmic- self compiled (do not seem
to be able to compile deb source package with ilbc, and deb package
does not have ilbc)

Any idea what may be happening?

John

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