IAX2 Client to PJSIP DIAL

Asterisk Asterisk Endpoints
1

I have a very weird problem

I have a client connected to Asterisk Box from the cloud via IAX2 then it dial out to a Cell via PJSIP with the following…
Dial(PJSIP/1${EXTEN:1}@anveodirect_endpoint,30,r)

Funny thing is the call goes through but no two way audio ever
but lets say the client calls a business like cable company at&t customer care, there is two way audio???

2

It appears to be glitchy when going from IAX2 to PJSIP
Is anyone else getting similar problem or am I missing something?

3

I haven’t heard of any reported problems like that, but not many individuals use IAX2. You’d need to provide debug information to see where the problem is - “pjsip set logger on” and “rtp set debug on” in the case of PJSIP which will show the signaling and the media flow. Information about the environment is also useful, for example if you are behind NAT and PJSIP has not been configured to know this then depending on the remote side audio problems can occur.

1 Like
4

I found the issue, I was using my Cell Phone Data as a Test softphone (zoiper) to test outside my network. Apparently my Cell Data is blocking specific ports. My cell data will block all numbers and not business numbers liek att or tmobile customer service numbers. I thought thats was interesting.