[solved- was pjsip codec setting] Suddenly not sending any audio on dev server.

ghost commented 4 years ago

Two servers, same versions of Ubuntu, Asterisk, same configs - same everything as far as I can see.

It's just that the dev server now rtp bridge doesn't seem to hear any audio (I set my own custom payload for a timeout - but it's working on the other server!

MQTT is working, too - I realise this isn't your problem but any ideas on diagnosing externalMedia channels? Thanks! Troublesome log below - on the working server, it has the proper full-on conversation with Dialogflow

0|rtp-server | [1588783673320] INFO (Asterisk-Dialogflow-ARI-Bridge/27648 on ip-x-x-x-x): created an externalMedia channel 0|rtp-server | [1588783678633] INFO (Asterisk-Dialogflow-ARI-Bridge/27648 on ip-x-x-x-x): got a message 0|rtp-server | topic: "dialogflow-asterisk/1588783667.90/events" 0|rtp-server | payload: { 0|rtp-server | "timeout": true 0|rtp-server | }

danjenkins commented 4 years ago

I'd guess youre not getting the RTP... so first thing to check is, are you getting packets on your port you've specified for media? tcpdump should tell you :)

ghost commented 4 years ago

Hi @danjenkins FOUND IT! Thank you so much for suggestion tcpdump.

And after half a day, the culprit was as simple as this, although I have no idea why, but it might help others...

In pjsip_wizard.conf: ~endpoint/allow = alll~ endpoint/allow = alaw

Before this, I was seeing nothing on 7777 via sudo tcpdump -i lo port 7777 -vv -X

Nothing in Asterisk via rtp set debug on

And the build of EVERYTHING was the same on both machines, and rtp show settings was identical


General Settings:
----------------
  Port start:      10000
  Port end:        20000
  Checksums:       Yes
  DTMF Timeout:    1200
  Strict RTP:      Yes
  Probation:       4 frames
  ICE support:     Yes

danjenkins commented 4 years ago

Is that alll or all ?

On Thu, 7 May 2020, 14:54 digitaltoast, notifications@github.com wrote:

Hi @danjenkins https://github.com/danjenkins FOUND IT! Thank you so much for suggestion tcpdump.

And after half a day, the culprit was as simple as this, although I have no idea why, but it might help others...

In pjsip_wizard.conf: endpoint/allow = alll endpoint/allow = alaw

Before this, I was seeing nothing on 7777 via sudo tcpdump -i lo port 7777 -vv -X

Nothing in Asterisk via rtp set debug on

And the build of EVERYTHING was the same on both machines, and rtp show settings was identical

General Settings:

Port start: 10000 Port end: 20000 Checksums: Yes DTMF Timeout: 1200 Strict RTP: Yes Probation: 4 frames ICE support: Yes

— You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub https://github.com/nimbleape/asterisk-dialogflow-rtp-audioserver/issues/16#issuecomment-625269904, or unsubscribe https://github.com/notifications/unsubscribe-auth/AAB3LLLRV5MU2JQ2MT5CDX3RQK4QLANCNFSM4M2TOVPA .

ghost commented 4 years ago

Ah, good spot - not sure how that crept it - it was definitely all, with 2 ells!

So changing from all to alaw fixed it. Asterisk boggles my mind daily...

nimbleape / asterisk-dialogflow-rtp-audioserver

[solved- was pjsip codec setting] Suddenly not sending any audio on dev server. #16

General Settings: