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/ ---------------------------------------------------------------------------
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. -----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/ --------------------------------------------------------------------------- _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
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/ --------------------------------------------------------------------------- _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
--------------------------------------------------------------------------- Peter Beckman Internet Guy beckman at angryox.com http://www.angryox.com/ ---------------------------------------------------------------------------
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