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
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