[VoiceOps] Asterisk Inconsistently sending RTP for deterministic set of orig DIDs

[Peter Beckman]

I'm dealing with a strange situation and I'm hoping there might be someone
who can see an easy answer.

1. When a call comes into Asterisk, we answer the call and send a SIP 200
     OK. Then we play an audio clip, then bridge the call to a Dial() with the
     'r' option to play ringing to the origination side of the call. This
     works with all of Carrier B's DIDs and most of Carrier A's DIDs. The
     RTP streams start sending as soon as we answer() the inbound call.

2. For other DIDs, we answer() and send a 200 OK and do not play any audio
     but bridge the call directly to a Dial(). The audio is still passed to
     the caller and they hear ringing generated from Asterisk, not locally.

3. For the problem DIDs we are working on right now, they all look like #2
     but no audio is passed to the caller, and the RTP stream is not sent
     (based on tcpdumps).  However when the Dial()ed call leg answers, the
     RTP stream begins (inconsistently, but that's another issue).

Because this issue is happening with just a small subset of DIDs on one
specific carrier of ours, and that we have made NO changes to our Asterisk
configuration or our AGI that handles calls, and that the same AGI handles
all inbound calls the same way, I'm looking for any troubleshooting advice
I can find.

This started very suddenly after several years of no issues three days ago.

My Asterisk server has enough inodes, very few open files (no where close
to ulimit levels), and no indication that there are any problems. There are
no limitations on inbound or outbound ports for RTP (no firewall rules
restricting that traffic). Reinvites are not enabled.

Beckman
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Peter Beckman                                                  Internet Guy
beckman at angryox.com                                 http://www.angryox.com/
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