Brent Torrenga wrote:
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?
I have the same problem, only my customers report that it only happens
occasionally. Most of the time, they can transfer calls just fine.
They can also put calls on hold and retrieve them as expected. However,
sometimes, about once a day, they try to recover a call and the caller
can't hear them, but they can hear the caller.
I've seen this happen once, but I've been unable to reproduce it reliably.
Any ideas?
Mike Diehl.
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