I’ve recently deployed an HT813 directly locally linked to a FreePBX (Asterisk 19.3.2). HT813 is connected as a trunk to the Asterisk, so the registration is deactivated. I have an outbound route configured in the Asterisk to hand over the calls with this dial pattern to HT813. Incoming calls (Analog line from FXO to any VOIP extension) work perfectly.
For the outbound calls, when any VOIP Extension dials to an external (analog) extension, it’s properly handled to HT813 but Grandstream is taking ~8s to initiate the dial/ringing on the analog line.
Checking on FreePBX logs we have:
[2022-12-30 18:08:31] VERBOSE[C-00000010] app_dial.c: Called PJSIP/95@Grandstream
[2022-12-30 18:08:39] VERBOSE[C-00000010] app_dial.c: PJSIP/Grandstream-00000022 is ringing
Full PJSIP Logs here: https://pastebin.com/LxjiQiGy
I’ve read that these delays generally are caused by DNS resolution, but I do have the Asterisk hostname on the hosts file. Also, to my naive eyes, it doesn’t seem like an Asterisk fault as in the logs we can see the 8s delay between Grandstream sending the “SIP/2.0 100 Trying” and the “SIP/2.0 180 Ringing” (Actually ring).
My infrastructure and issue seem very similar to this issue: https://forums.grandstream.com/t/ht813-dial-delay/34404
Has anyone ever seen this scenario and has a clue which configuration I’m missing here?
Thanks a lot and Happy Holidays!