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=
mailbox=3001
context = sip
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:
Looks like I did not check what codes are supported on SJLABS.
It does not support g726.Thats why it is not working.
I checked it with gsm and is working.
My fault.
Sorry
Hi,
I would like to know if asterisk is able to translate between two
differnet codecs. For example:
I have this config
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
