I’m trying to connect two Asterisk servers in the same LAN: one is v. 16 and the other is v. 1.4.
I am progressively moving the user extensions/endpoints from 1.4 to 16, but I need a SIP trunk between the two servers so that calls are made both ways (ie. I want extensions in 1.4 to able to call extensions in 16 and vice versa).
I tried setting up a “friend” trunk without authentication, but it didn’t work out.
So I tried the following configuration which works for users calling from 16 to 1.4.
This is on the v. 16 server:
[trunk_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/allow = !all,alaw,ulaw,opus,gsm,vp8,h264 aor/qualify_frequency = 30 registration/expiration = 1800 [meetbox](trunk_defaults) sends_auth = yes accepts_auth = yes sends_registrations = yes accepts_registrations = yes remote_hosts = 10.215.147.115 endpoint/context = custom-newsystem endpoint/from_user = meetbox outbound_auth/username = meetbox outbound_auth/password = xxxx inbound_auth/username = meetbox inbound_auth/password = xxxx
this is on the v. 1.4 server:
[meetbox] disallow=all type=friend host=10.215.144.92 username=meetbox secret=xxxx qualify=yes insecure=invite allow=alaw allow=ulaw allow=gsm allow=opus allow=vp8 allow=h264 context=custom-newsystem videosupport=yes
However, if users from v. 1.4 try to call anything in v. 16, I get several of these in the CLI of v. 16:
NOTICE: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '"My Test" <sip:email@example.com>' failed for '10.215.147.115:5060' (callid: firstname.lastname@example.org) - No matching endpoint found NOTICE: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '"My Test" <sip:email@example.com>' failed for '10.215.147.115:5060' (callid: firstname.lastname@example.org) - Failed to authenticate
What does “No matching endpoint found” mean? When this happened the SIP user in v. 1.4 was trying to call sip:email@example.com where “4053” exists as a PJSIP endpoint in v. 16.
Does “endpoint” in this case refer to ‘313537021’, and is Asterisk trying to find it in v. 16? Why? (I mean, why does a “caller” have to exist there?)
In these cases is it necessary to add a line such as the following in sip.conf?
It doesn’t seem to be.
If I do, here is what I see on v. 16 after adding ‘aor/max_contacts = 1000’ in pjsip_wizard:
Endpoint: meetbox Not in use 0 of inf OutAuth: meetbox-oauth/meetbox InAuth: meetbox-iauth/meetbox Aor: meetbox 1000 Contact: meetbox/sip:10.215.147.115 92e3d71c31 Avail 2.584 Transport: transport-udp udp 0 0 0.0.0.0:5060
Still, I’m getting the same behavior for “incoming” calls on v. 16 (No matching endpoint found).