From what I can see....
The 2 legs of the call are: 'phone' in alaw and 'laptop' in g726, why
should he need G.729 anywhere ?
Bartosz, not exactly that familiar, but I guess you could try to debug
the call establishmment.
(one thing that puzzles me, you mention "IAXy", but you show 2 sip.conf
entries..should not be one in iax.conf and one in sip.conf ?
(of course, with the proper syntax for each .conf file).
Moises Silva wrote:
be sure you allow the g729 codec in [general] context in sip.conf for
the sjphone.
On 1/2/06, Bartosz Wegrzyn - asterisk <[EMAIL PROTECTED]> wrote:
Hi,
I would like to know if asterisk is able to translate between two
differnet codecs. For example:
I have this config in sip.conf file:
[phone]
disallow=all
allow=ulaw
dtmfmode=rfc2833
dtmf=rfc2833
username=phone
type=friend
host=dynamic
secret=xxxx
mailbox=3001
context = sip
callerid="Wireless <3001>"
canreinvite=no
qualify=yes
qualify=3000
nat=yes
[laptop]
disallow=all
allow=g726
dtmfmode=rfc2833
dtmf=rfc2833
username=laptop
type=friend
host=dynamic
secret=xxxx
mailbox=3002
context = sip
callerid="Laptop" <3002>
canreinvite=no
qualify=yes
qualify=3000
nat=yes
Should asterisk translate between two codes.
First clent is iaxy, second is sjphone.
It is not working for me, and I am getting error on sjphone:
"Unabke to agree on media streems".
When I change the codec for laptop to ulaw everything worls ok.
This would mean that asterisk cannot establish communication if both ends
have different codecs supported. Is this right???
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