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 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. NAT is notorious for causing one-way audio and transcoding G.729 requires a commercial license from Digium. 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? I don't use either format, but I believe the required modules are "format_g729.so" and "format_pcm.so". Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer
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