[asterisk-users] SIP??
Florea Igor
igor.florea at topex.ro
Fri Feb 9 02:03:11 MST 2007
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,
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
--
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
Local time: GMT+2
www.topex.ro
More information about the asterisk-users
mailing list