We have recently had a “short notice” move for our PBX. Originally running with an E1/PRI which was operating fine, we have had to move to SIP trunks in the interim while our PRI is relocated.
After some issues we have the SIP Trunk registered. However ongoing issues with incoming and outgoing calls. Trunks have been checked and double checked by more than 1 person.
Below is the current config, if anyone can provide some feedback/suggestions it would be much appreciated:
OK issue is that we have a registered SIP Trunk (no issue there), all OK on that front.
However unable to make incoming or outgoing calls.
Both incoming and outgoing routes have been setup and are correct.
The PABX was operating with a PRI/E1 previously and all was working OK, so we know the routes were fine, changes only made to reflect the introduction of SIP.
PBX is sitting behind a firewall, which has UDP and TCP Ports forwarded as required.
Are there any specific changes that we should be making to asterisk that we may have overlooked for the SIP trunk??
When we make a call to the DID from an external phone/line, we receive a message “your call could not be completed at this time”, where as prior to the SIP Trunk registration the call would just “drop”.
From an internal perspective (calling out), we receive an “all circuits are busy” notification.
Can you open the CLI and watch while you try an inbound call? You should see SIP messages if the call is making it to your Asterisk server. If you see SIP messages with the calling number, you should be able to see what is failing. If you see no messages at all, then you have a problem with the trunk or route.
If you’re not sure how to connect and view the command line, just connect to your server via SSH (PuTTy), and then run ‘asterisk -rvvvv’.