[Asterisk-Users] Is SIP "canreinvite" working ok?

[Álvaro Palma]

I've the following situation:

Phone A: Codec GSM supported
Phone B: Codec iLBC supported

in sip.conf:

[general]
...
disallow=all
allow=gsm
allow=ilbc
allow=alaw
allow=ulaw
canreinvite=yes
...

(There's a lot of other SIP users, that's why I made the default codec 
list bigger than just GSM and/or ALAW)

If phone A calls to phone B the conversation is established at SIP 
level, but there's no RTP traffic between the machines. If I make a "sip 
show channels at the Asterisk console, I see:

server*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold 
Last Message
192.168.1.101    phone_A     10095d01445  00103/00000  ulaw  No 
Tx: ACK
192.168.1.107    phone_B     182E175F-F6  00102/00002  ulaw  No 
Tx: ACK
2 active SIP channels

(ULAW?!?!?, not even ALAW!!!)

As far as I understand, since in this case the communication can not be 
established directly between A and B (i.e. bypassing Asterisk as the 
media transport), given the fact that the A codec and the B codec are 
different, the REINVITE shouldn't be issued and discarded automatically 
for Asterisk, even and despite the fact canreinvite=yes is set. However, 
it seems to be issued anyway, so I can't hear anything.

Am I doing something wrong, or this is effectively an Asterisk problem?

Asterisk 1.2.4
SIP Client: SJPhone 1.60.289a

I checked the REINVITE sent from Asterisk to the phones with Ethereal.
Also, if I set canreinvite=no, the communication works nice, with GSM 
for one side and iLBC in the other.

Thanks a lot for your attention.

-- 
Atly.
Alvaro Palma