[asterisk-users] SIP??

[Vicky]

config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer
definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc )

On 08/02/07, Florea Igor <igor.florea at topex.ro> wrote:
>
> Hi,
> I'm new to *,so i apologize for stupid questions.
> I'm having problem with this arhitecture:
> I'm calling asterisk from behind a NAT(sjphone user) with a low band so
> I'm
> using GSM codec.
> In extensions.conf I have:
> exten => 337,1,Dial(SIP/99@<ip_pbx2>)
> so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2.
> RTP stream between sjphone and Asterisk are ok (GSM).
> The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although
> ip_pbx2 sip is telling asterisk It only knows "codec 0"
> Is this a config problem or a bug?
> Igor,
>
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