[asterisk-users] canreinvite=yes and RTP dropping in and out

[Gary Richardson]

Hey guys,

I'm having yet another strange problem. I've recently set canreinvite=yes,
allowing the RTP streams to avoid our * server. Now, a few people are
experience one way audio drops on internal calls. External calls are working
fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20
seconds or more, the stream will resume. Flipping the person on and off hold
won't resume the stream.

We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem
to happen all of the time. There are no sip messages being exchanged when
the stream stops or restarts.

Any suggestions?

Thanks.
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