I have recently configured FreePBX with Grandstream GXW 4014 FXO gateway using for now just one analong PSTN line.
I have configured following along Configuring a Grandstream GXW-410X Device to act as an FXO Gateway, but I can’t make any outbound call, I have even connect old analog phone directly to wall inlet, and tested few times, just to confirm that everything is OK. Everything I get in my handset is “all circuits are busy now please try again later”. Just note, I have Yealink T41P VoIP telephones, and custom made server with latest FreePBX 13.9.1 version and GrandStream GXW 4104 FXO gateway with latest Program : 188.8.131.52, Loader : 184.108.40.206 and Boot :220.127.116.11 version.
My configuration is as follows:
Calling to VoIP >> Calling to VoIP Unconditional Call Forward to Following:
Outgoing dial plan (Stage 1 dialing in GXW) is as following:
User account is set up in GXW4104 as follows:
1 101 101 somepassword Account 1
Trunk name in FreePBX is “101” as account set up in GWX, tech is “sip”, and Dial Number Manipulation Rules is just a dot sign “.” to allow all digits and characters.
My “peer” details are:
My intbound route is set with my number as DID (“38515626012”). and destinastion is set as extension (200).
My outbound route is set with CID and route name “101”, where Trunk Sequence for Matched Routes is “101”.
Dial pattern is set to single dot character “.”.
I have tried various changes, and have eliminated network related problems. I have also took a look in /var/log/asterisk/full and then seen something problematic, but I don’t have idea what to check, or where to look.
[2016-08-18 00:01:37] NOTICE[C-0000000f] chan_sip.c: Failed to authenticate on INVITE to sip:[email protected]:5160;tag=as5a62a413
Here is a part of asterisk log, when I try to dial my phone number, I haven’t put it all here because is to big.
**NOTE: I have emptied log before making test call, which than generate entries in log file.