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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