Trouble connecting a SIP call

[Dave Kovalcik]

Cant get SIP working - - system says Established but won't dial a number to a cellphone - using version 5.2 on Centos - all ports open - what log files do you need to see - and is there a testing SIP provider that you recommend?

 

[Max]

Good day.
Please clarify the following:
1. What WCS version do you use?
2. What example application do you use?
3. Is the problem reproduced in the same example application on our demo server https://demo.flashphoner.com:8444/?
Also, prepare a report as described here and send to support@flashphoner.com
As for testing, we recommend to install your own OpenSIPS or Asterisk instance to test SIP calls in environment you can manage.

 

[Dave Kovalcik]

We just sent our logs to you - we are trying to use SIP addresses as well as phone numbers to create an RTMP bridge - thanks for the help -

 

[Max]

Good day.
In logs you've sent we see 2 tries to establish a SIP call:
1. Your first call was not established because you probably set wrong authentication parameters, here's SIP signaling sequence from server log:

--------------------> INVITE sip:callee@domain SIP/2.0
<-------------------- SIP/2.0 407 Proxy Authentication Required
--------------------> ACK sip:callee@domain SIP/2.0
--------------------> INVITE sip:callee@domain SIP/2.0 with digest authentication data
<-------------------- SIP/2.0 403 Forbidden

2. Your second call was not established because you set wrong callee: SIP login like "callee@domain" is not allowed.
So you should check your SIP authentication data.

 

[Dave Kovalcik]

I got the system to work last night but I was under the impression that we could connect to a zoom session via a sip@domain.com address ? Is this not possible?

Ultimately we are trying to connect telepresence devices to an RTMP wowza streaming solution. They use h264 via a sip audio/video connection.

Can this be done and if so how?

 

[Dave Kovalcik]

We have been using this software for over 2 years - this is a new request from our client - I posted another question similiar to this - can we do this through zoom??? in other words have our client connect to zoom and then capture the zoom video / audio feed and bridge that to rtmp?

 

[Max]

Please see this doc: republishing a SIP call to an RTMP stream
As to using SIP URI for callee, the format is 'sip:username@domain'.