[Asterisk-Users] Translating between different codes
Julio Arruda
jarruda-asterisk at jarruda.com
Mon Jan 2 08:42:16 MST 2006
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 <junk at lexoncom.com> 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???
More information about the asterisk-users
mailing list