On Tue, Jul 19, 2011 at 3:46 PM, Matthew J. Roth <[email protected]> wrote:
> Michael, > > Here are the differences between the systems that I determined from the two > SIP traces: > > * Working system: no NAT, phone codec: G.729, Asterisk codec: G.729 > * Non-working system: NAT, phone codec: G.729, Asterisk codec: A-law > 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. > > Does the conversation have two-way audio prior to the hold? If it doesn't, > your problem may be caused by NAT and/or codec transcoding. > Thr call is fine prior to the HOLD. > NAT is notorious for causing one-way audio and transcoding G.729 requires a > commercial license from Digium. > G.729 licenses have been purchased from Digium and installed. We tested with NAT set to "no" and "yes" and neither settings mattered. > > After those two possible causes are ruled out, the only other thing that I > can think of is a missing format translation path for the music-on-hold > files. Does the AsteriskNOW system have modules loaded for both formats? > It should. How do we check this? > I don't use either format, but I believe the required modules are > "format_g729.so" and "format_pcm.so". > Thanks.
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