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

[Peter Beckman]

On Thu, 19 Jun 2014, Eric Wieling wrote:

> Some people, when confronted with a ringback problem, think "I know, I'll
> use the 'r' option to Dial."   Now they have two problems.  -- with
> apologies to Jamie Zawinski
>
> I don't think the 'r' option to Dial has solved anything for anyone in
> many years.  Asterisk will normally generate ringback if it thinks it
> should, such as during a Dial.   If that doesn't work, try using the
> Ringing or Playtones app.   All 'r' does is unconditionally block early
> media from the destination and replace it with generated ringback audio.

  Using 1.4.28.

  The biggest issue is that all of our carriers behave inconsistently to
  signaling, and with our solution we sometimes play a short audio clip
  prior to forwarding the call. If we don't answer() the call, the different
  carriers may or may not play that audio prior to ringing.

  Got any pointers to Asterisk versions and how to unify and solve
  inconsistent carrier behavior without the global 'r' hack? I'll go Google
  it, but if you have any you can share, thanks!

Beckman


> -----Original Message-----
> From: VoiceOps [mailto:voiceops-bounces at voiceops.org] On Behalf Of Peter Beckman
> Sent: Thursday, June 19, 2014 2:58 PM
> To: VoiceOps
> Subject: [VoiceOps] Asterisk Inconsistently sending RTP for deterministic set of orig DIDs
>
> 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
> ---------------------------------------------------------------------------
> Peter Beckman                                                  Internet Guy
> beckman at angryox.com                                 http://www.angryox.com/
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Peter Beckman                                                  Internet Guy
beckman at angryox.com                                 http://www.angryox.com/
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