I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though the local phone can hear the Sipphone user.
Every possible combination of nat=yes, no, maybe, possibly or never gives the same result. Further, canreinvite=yes/no/nonat has no result. I suspect a possible reinvite issue with Asterisk being out of the RTP stream, so I have tried all the usual variables in the DialI() command as well to no avail. Any thoughts on how to fix one-way-audio after a hold? --Brent
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