Asterisk with Webrtc JSSIP ACK null

Asterisk Asterisk WebRTC
1

I’m trying to build a softphone with angular 11.

but having problem with asterisk with webrtc softphone.

It’s dropping a call after time between 15s and 90s.

I’m using in server side Asterisk v16.19.1 With FreePBX v15.0.17.43 Hosted on Gcloud compute engine.

In client side I’m using Jssip v3.8.0

Log from Asterisk: Asterisk Log - Pastebin.com

Log from jssip: Jssip log - Pastebin.com

I logged the trigger “connected” on jssip and get this:

phone-bar.component.ts:182 -------------> {originator: "local", ack: null}
2

When debugging on Asterisk, you generally want at least verbosity 5. Debug 5 may also be desirable.

Unfortunately, this means you will get a very large log, because of the amount of dialplan that FreePBX executes, so you need to try and look for likely problem areas before you upload it as people here are not used to ploughing through FreePBX dialplan, which they will not fully understand.

At the moment there is no evidence that the cause is FreePBX related, so I haven’t redirected you to the FreePBX forum. However, if you can reproduce the problem on bare Asterisk, with a minimal dialplan (e.g. just Dial(), it would be much easier to debug.

3

Hello david, I’m not sure what I did wrong.
but I reinstalled asterisk with freepbx from this tuto:
http://www.osslab.tw/books/asterisk-freepbx/page/install-freepbx-15-with-asterisk-16-on-debian-10

and then this one for webrtc:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

and now it works!

Thanks for your patience!!

4

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