Hi everyone,
I have been really struggling for the registration of the trunk for both IP and credential based authenticaion.
Anyone can please share any documentation or guide to configure it.
Have a look at the link below:
That’s through the API, I need it by the WEB UI
I think your problem is probably with the codec. Asterisk made a change a few versions back on the allow command and removed the disallow command.
Actually it not just the codecs, I have done the registration. The trunk is registered but still getting the following error.
ERROR[582228]: res_pjsip_outbound_authenticator_digest.c:450 digest_create_request_with_auth: Host: ‘SERVER IP HERE:5060’: There were no auth ids available.
and when I try to make call between 2 softphones that are registered using the lines, the call gets disconnect exaclty after 32s.
It would be of great help if anyone can share a screenshot of the lines and the trunk configrations and I can use it according to my use.
Hello,
If your SIP trunk won’t REGISTER until you make an outbound call (INVITE,401,INVITE), consider using “line = yes” in your registration trunk tab.
I saw this behavior starting 22.X version (don’t remember which one)
Wazo will send REGISTER packets with this option.
Does it solve your issue ?
Here is my Screenshots this is IP authentication.
have this line = yes, actually trunk is registered but I am stuck at the problem of Everyone is busy when I try to make the call.
let me share you the SIP traces
I have created 2 users and their lines, and trying to register them on two softphones and trying to call. the call is coming fine but just without the voice.
can anyone please help in debugging it, SIP traces and configurations are attached
In your SDP, the IP address is private, so if your are natted you need to configure correctly the Nat. Please check out documentation for the Nat.
I don’t see any information for your outgoing call. Line = yes is actually for multi-tenant in general for wazo, it just permit to have a line id in the registration to permit to find the good information of the invite from your sip provider. It’s not always supported.
Have you created the rules for an outgoing calls via your sip provider?
Finally got the issue sorted, both outbound and inbound is working fine. Thank you so much for the support.