ricardojlrufino / webphone-sip

WebRTC SIP based VoIP client software (+chrome extension)
https://ricardojlrufino.github.io/webphone-sip/
99 stars 58 forks source link

Not incoming calls on Asterisk #10

Closed uribes78 closed 4 months ago

uribes78 commented 4 months ago

This is more a question than a request issue.

I forked you repo and modified it to use it as popup instead of a new window, looks like this:

imagen

and once the config is set, it can register without problem, looking like:

imagen

If I make a outbound call goes fine, no problem. Event if I close the popup its continue on call and if I open de popup render as need.

The problem I have is with incoming calls. I can't received the INVITE from asterisk. What I saw in asterisk is that the peer show UNREACHABLE on the qualify parameter.

Do you have an advice about it? Did you try to use it like this? Hope you can give a hint. Thanks

uribes78 commented 4 months ago

Hi @ricardojlrufino , I have solved the issue. Reading many forums and documentation (even asterisk chan sip code), I change the transport peer parameter, leaving as transport=ws,udp, for some reason you need to have both to let core asterisk works otherwise you may see the message:

ERROR[880980]: chan_sip.c:4354 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

Once applied changes the webphone can receive incoming calls, as you can see in this image:

imagen

Thanks for sharing your application.