Hi Guys, Tried the disallow=all and allow=all but still getting one way audio with x-lite and messenger.
Any update on this problem. Dave ----- Original Message ----- From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, October 24, 2003 1:49 PM Subject: Re: [Asterisk-Users] Tonight's CVS breaks Grandstream phone > >FYI. Haven't dug enough to be able to report any more, but > >re-fetched CVS to verify that sometime in the last few days CVS > >changes now break my GS phone. > > > >It appears to be at the RTP level. It seems to set the call up just > >fine, but no audio is passed back to the instrument. > > > >I reverted, and will try to play with this tomorrow unless someone > >else tells us it's fixed. > > > >Thx. > > > >B. > > I am seeing the same error with CVS as of 02:00 today GMT. > Grandstream phones will dial, the dialplan will seem to work, but > after a few seconds the call fails. Looking at the SIP debug, I see > that > > There was a new feature added last night to allow for codec > permission/denial on a per-peer basis in sip.conf. This means that > each SIP client can be forced to use particular codecs (at least, > that is the intent. more testing, anyone?) > > So, it seems that the Grandstreams do not elegantly handle some > circumstances of codec presentation which were created by these new > patches. It is necessary for you to put the following lines in each > Grandstream entry in your sip.conf, OR you can put the identical > entries in [general] to have it work across all clients. Note that > both lines are required: > > disallow=all > allow=all > > > JT > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
