Yep Cisco ATA had to RX TX and LBR had to be set to 3 ----- Original Message ----- From: "Brian West" <[EMAIL PROTECTED]> To: "Asterisk Developers Mailing List" <[email protected]> Sent: Thursday, April 07, 2005 12:38 PM Subject: Re: [Asterisk-Dev] G729 Question
> I think we already covered this in the dev conf didn't we? > > /b > > On Apr 7, 2005, at 2:21 PM, Tom Dickenson wrote: > > > SOMEONE WANT TO EXPLAIN THIS? While finally placing an outgoing call > > using > > mutualphone (requiring that I get a G.729 License to do this) the > > following > > pops up on asterisk's cli when a call is made (call works wondeful) > > just > > curious to the extent of the following lines... > > > > Attempting native bridge on SIP/cisco1-5d0b and SIP/mutualphone-c59c > > > > Apr 7 12:01:58 WARNING[726]: rtp.c:1559 ast_rtp_bridge: codec0 = 12 > > is not > > codec1 = 256, cannot native bridge > > > > -Tom Dickenson > > > > _______________________________________________ > > Asterisk-Dev mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > _______________________________________________ > Asterisk-Dev mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
