I have everything set up in terms of a phone, extension.
This is the error log I’m receiving from the Asterisk CLI when trying to make a call
Connected to Asterisk 13.0.2 currently running on localhost (pid = 3080)
[2014-12-19 16:35:53] WARNING[3134]: func_cdr.c:349 cdr_write_callback: CDR requires a value (CDR(variable)=value)
[2014-12-19 16:35:53] WARNING[4688][C-00000003]: chan_dahdi.c:13629 determine_starting_point: Unable to determine group for data r/07983497629
[2014-12-19 16:35:53] WARNING[4688][C-00000003]: app_dial.c:2431 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 0 - Unknown)
[2014-12-19 16:35:55] WARNING[4688][C-00000003]: app_playback.c:493 playback_exec: Playback failed on PJSIP/1105-00000003 for all-circuits-busy-now&pls-try-call-later, noanswer
As it says in the log, it always says ‘All Circuits are busy now…’
check with your origination/termination provider to ensure that traffic is being sent. i had the same issue and took me few hours to figure it out the first time when i first started with freepbx.
make sure your trunks are setup correctly and the IP of your orgination/termination provider is setup under Trunk > Peer Details > Host
make sure your pbx’s IP address is accepting/sending traffic from orig./termin. provider
make sure you have outbound routes configured and you have dial plans setup too.
@klrgirish198117 - Because this problem can have multiple causes, I strongly recommend you start your own thread instead of hijacking Mohammed’s thread. Additionally, as a general rule, please try and refrain from posting twice in a row; instead, use the edit function.
I guest you made a mistake on Trunk settings, as these settings it refer to wide trunks sources as, DAHDI, SIP IAX and others.
It seems on trace system it try to route to DAHDI channels (DAHDI channels it refer to TDM interfaces, as analog ports FXO and digital ISDN E1 PSTN trunks), it seems you don’t have any DAHDI resources installed on you system only SIP?
Please check BSNL trunks setting to which resource have been set, good luck
I recently had an interesting experience, while trying to make an international call I was getting the same thing “All circuits are busy right now”.
This is my finding on this, not sure how it happened but it must have been with the last Sipstation module update, looking at my routes, a new route is in there called Sipstation-Int with an international dial prefix of 011 inside it.
It was ahead of my main International dial trunk for FlowRoute, I am not sure how this happened as I do not recall adding the route nor placing ahead of others.
My point being, check your routes, make sure your placement is correct and another is NOT picking up the dial prefixes or pattern matching, as this was my issue.
This is originally pointed to the OP… not that it matters, but as the guy above mentioned “others may need to start their own threads”.