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 ) <br><br><div><span class="gmail_quote">On 09/02/07, <b class="gmail_sendername">Florea Igor
</b> <<a href="mailto:igor.florea@topex.ro">igor.florea@topex.ro</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729
<br><br>On Thursday 08 February 2007 19:00, Vicky wrote:<br>> config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer<br>> definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc )<br>><br>
> On 08/02/07, Florea Igor <<a href="mailto:igor.florea@topex.ro">igor.florea@topex.ro</a>> wrote:<br>> > Hi,<br>> > I'm new to *,so i apologize for stupid questions.<br>> > I'm having problem with this arhitecture:
<br>> > I'm calling asterisk from behind a NAT(sjphone user) with a low band so<br>> > I'm<br>> > using GSM codec.<br>> > In extensions.conf I have:<br>> > exten => 337,1,Dial(SIP/99@<ip_pbx2>)
<br>> > so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2.<br>> > RTP stream between sjphone and Asterisk are ok (GSM).<br>> > The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although
<br>> > ip_pbx2 sip is telling asterisk It only knows "codec 0"<br>> > Is this a config problem or a bug?<br>> > Igor,<br>> ><br>> > _______________________________________________<br>
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<a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br>--<br>There are 10 kinds of people in the world: those who know binary and those who<br>
don't.<br><br>Igor Florea<br>Ing. dezvoltare<br>Phone: +40 21 232 04 24<br>Fax: +40 21 232 31 56<br>Local time: GMT+2<br><a href="http://www.topex.ro">www.topex.ro</a><br><br>_______________________________________________
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