Michael wrote: > > True. In the working system, LAN calls are also using G.729, while > in the non-working system, LAN calls are in G.711 (supported but > not prioritized by the phones) and only the SIP trunk to the ITSP > is set to G.729.
Can you set the phone to G.711 and try making a LAN call on the non- working system. If a call that is G.711 from end-to-end doesn't have the same problem it would be evidence of a codec translation issue. > We tested with NAT set to "no" and "yes" and neither settings > mattered. As long as the phone and the Asterisk server are both on the same LAN, my recommendation would be to test with NAT set to "no". NAT is not necessary unless there is a firewall between the phone and the Asterisk server and setting it to "no" also eliminates a variable that differentiates it from the working system. > It should (have modules loaded for both formats). How do we check > this? The following command should output a line for each module (as shown): # asterisk -rx 'module show' | egrep 'format_g729.so|format_pcm.so' format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 5 format_g729.so Raw G729 data 0 Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
