[asterisk-dev] Change of Codec For Blind Transfer

[Chan Kwang Mien]

Hi,

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


sip1 supports g711 only
sip2 supports g711 and g729 only
sip3 supports g729 only

sip1 establishes a call to sip2. sip1 then executes a blind transfer so
that sip2 will be connected to sip3.

>From the sip logs, Asterisk received a REFER message from sip1. It then
sent an INVITE with codec=g729 to sip3 to set up a call to sip3. When
Asterisk tried to bridge the call between sip2 and sip3, it couldn't
because sip2 is still using g711 while sip3 uses g729. The call is
dropped.

Shouldn't Asterisk send a Re-INVITE to sip2 to force sip2 to change its
codec to g729 ? In this way, sip2 can talk to sip3 using g729.