All Curcuits Are Busy Now


#1

Hi Guys, I have an issue with my outgoing routes. I can receive calls but when I go to make an outgoing call it fails with “all circuits are busy now please try your call again later”. Any help would be appreciated Please see my log files below.

https://pastebin.pl/view/7aac5888


#2

Lines 190 - 191:

[2021-09-15 10:49:47] VERBOSE[2296][C-0000001d] app_dial.c: Called SIP/0280905394/0458001199
[2021-09-15 10:49:47] WARNING[2588][C-0000001d] chan_sip.c: Received response: "Forbidden" from '<sip:0280905394@192.168.0.14:5160>;tag=as70eb03c7'

The outbound route seems to be ok; it sent the call to the trunk named 0280905394. However, that trunk, apparently at 192.168.0.14, rejected the call. Is that the correct address? What device do you have at that address (a POTS, PRI or GSM gateway? another PBX? an SBC to a VoIP provider?) Possibly, that device is not correctly configured, your trunk configuration is not sending the correct headers, or the destination number or caller ID is not in the format that the device expects.

Please provide details about the trunk settings and the device to which it connects.


#3

Hi Stewart1 I apreciate your reply. I am out of the office but will post first thing tomorrow morning. Again thank you for helping.


#4

Hi Stewart1 My pbx is located at 192.168.0.14. I have two trunks that are registered and one outbound route. We are using cisco phones and all extensions are registered. Do you need anything else?


#5

Sorry, I misinterpreted the Forbidden error; apparently Asterisk was reporting the From header in the 403 response.

The trunking provider rejected the call. Things to check:
Destination number in correct format.
Caller ID in correct format. Use the same format as you receive on an incoming call.
Your trunk configuration may need to set fromuser (to the same username as in your register string) and/or fromdomain (to the same value as you have for host). Check your provider’s documentation.

If you still have trouble, at the Asterisk command prompt type
sip set debug on
make a new test call and paste the Asterisk log for the call, which will now include a SIP trace.


#6

Hi Stewart1 thanks for the help! I will look at the system again and let you know how I go.


(Lisa Purchasing) #7

I am a newbie at FreePBX, but our former PBX tech left notes behind. Apparently they had this problem, too. His answer was: “That ‘All users’ directory is not going to be updated automatically when we create new users. That will be something to keep in mind. Go to User, Advanced tab, Default Group Inclusion, Change it to Include.”

Another note he left was about getting that same response when an IVR was set to play a recording but there was no recording. Once he uploaded and selected a recording, the calls went through.


#8

Hi Stewart1. Thanks again for the help, everything is now up and running. Thanks mate!!!