Hello,I need to use a SIP account to receive calls on an extension of my Asterisk. I am using the AsterisNow, and set up through the FreePBX.
If I set up my account at any Voip device I can receive calls without problems, but with account configured on Asterik not receive calls. When making calls with SIP same account I have no problems, only to receive is the problem.
Then describe the configuration as I have in the form of FreePBX, the option Trunks:
I clarify that I test with a route incoming calls to take all regardless of the DID to rule out a problem at that stage.
I have set in NAT Settings for SIP, my public IP and the local network.
Look SIP communications that come to my server and nothing appears when I call.
The trunk is recorded in this way:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
enter/user 77.72.169.134 No No 5060 Unmonitored
entrante/user 77.72.169.129 No No 5060 OK (390 ms)
Thanks for your help Hv, but still can not make incoming calls. Here is the record of the outgoing call. If this is not what you ask me, if not so tell me how it’s done ?.
If you look in your freepbx | reports | asterisk logs menu.
Check the log for a call you tried to receive.
in there you will find something like [2016-04-21 00:33:01] VERBOSE[30414] at the start
then use the command
grep “\[30414\]” /var/log/asterisk/full
replacing the 30414 with your own reference.
copy that paste it up and lets see what happens.
Hi Hv, and thanks again. Find a record, but even no record of incoming call, it was with an outgoing call, but not me of the code that you mention it.
The call never enters the asterisk from what I can see.
Any idea what I can do?
Check how you have setup the trunk.
If they are anything like sipgate leave the incomming section blank and just fill in the Peer stuff above it.
trunk Name
Peer Details and
Register string (below incoming).
Try and transplant your details into that config and see what happens
Just change the first 5 (FIVE) lines to your secrets.
Leave the rest as a test
and the register string looks ok BUT some like it to end with /user
I just looked at the VoipStunt website. Are you sure you can connect to them with Asterisk? Everything I read seems to indicate that they are an App based connection.
If they are a real VOIP provider, they should have instructions somewhere that tell you how to connect an Asterisk server to the system.