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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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 _______________________________________________ --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
_______________________________________________ --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
