[asterisk-users] Problem transferring calls between Cisco 7940 with SIP firmware

[Massimiliano Stucchi]

Hi all,

I'm having a strange problem with a bunch of cisco 7940G with SIP
firmware.  The problem arises when transferring a call coming in from a
SIP account to another phone.  The call connects, but for the first 10
seconds there is a situation with one-way audio, then it turns into a
fully working call.

I've googled extensively, but couldn't find much about this situation.
The problem is also present using the SCCP firmware, though.

All the phones are using a separate vlan with the server as well, so
there's no nat that has to be implied in here, and I can't understand
what's causing the problem.  What's more interesting is that if you send
any dtmf over from the phone which received the transferred call, the
one-way audio goes away and the call is fine immediately.

Here are some config files:

sip.conf

----

[902]
username=902
secret=****
context=prova
host=dynamic
type=friend
canreinvite=no
qualify=yes
nat=never
dtmfmode=rfc2833

----

I've already checked codec config and any other thing that could come to
my mind, but I'm getting out of ideas, so if anybody has any hint on how
to fix the issue, I'd be glad.

The problem is present both with asterisk 1.4 and 1.6, both latest
versions, and even with both sccp and sip on both versions.

If you need any debugging session, please ask and I'll provide.

Ciao

--

Max