[asterisk-users] SIP??

[Vicky]

check your sip.conf and make sure it has allow=ulaw and allow=alaw line (
you can even remove gsm to test it it works fine or not )

On 09/02/07, Florea Igor <igor.florea at topex.ro> wrote:
>
> ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729
>
> On Thursday 08 February 2007 19:00, Vicky wrote:
> > 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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> --
> There are 10 kinds of people in the world: those who know binary and those
> who
> don't.
>
> Igor Florea
> Ing. dezvoltare
> Phone: +40 21 232 04 24
> Fax: +40 21 232 31 56
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>
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