Then perhaps you have a NAT problem or some other issue. -- Tom
On 8/10/05, Gareth Blades <[EMAIL PROTECTED]> wrote: > I did try installing the 1.0.9 version but I have the same problem with > that release aswell. > > On Wed, 2005-08-10 at 14:14, Tom Hayden wrote: > > I encountered a similar problem with CVS-HEAD and sip2sip calls > > between our Polycom IP500s. I attempted to diagnose the problem and > > there are a few patches on mantis, but none of them worked for me. I > > flipped back to stable and have had no problems since. > > > > Anyone got any ideas? > > > > -- > > Tom > > > > On 8/10/05, Gareth Blades <[EMAIL PROTECTED]> wrote: > > > I am running the latest CVS version of Asterisk. > > > Calls between an IAX client and SIP phones (Grandstream SP2000 and > > > Sipura SPA-841) works fine and so do external call over the Internet > > > from the SIP desk phones. > > > > > > However when I call from either the Grandstream/Sipura phones to another > > > one I get no audio. I have the G711 ulaw codec defined as the preferred > > > on on all phones. > > > > > > Any idea what is going wrong? > > > I am guessing it is something to do with native transfers which is > > > performed in this situation. > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [email protected] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tom _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
