[asterisk-dev] Codec not changed when making an Attended xfer (REFER)

[Chan Kwang Mien]

sip1 <--> Asterisk <--> sip2
                ^
sip3 <-------|

sip1 supports g729 and g711 only
sip2 supports g729 and g711 only
sip3 supports g729 only
Asterisk-1.2.12.1 and Asterisk 1.4-beta2 used for the
testing


Test Scenario :

a) sip1 dials sip2 to establishe a call to sip2 using g711
b) sip.conf is changed such that sip2's preferred codec is g729
c) "sip reload" is executed at Asterisk
d) sip2 holds call
e) sip2 dials to sip3 to establish a call to sip3 using g729
f) sip2 presses "Transfer" button
g) Call between sip1 and sip3 is dropped

Asterisk complains that the codec used by sip1 and sip3 are incompatible,
thus dropping the call. This problem occurs whether canreinvite=yes or no

I was wondering if Asterisk could send an INVITE message to sip1 to change
its codec to g729 so that it could then talk to sip3.

May I know if there is any developer working on this problem ?

The bug# for this problem in the bugtracker is 7990.