Currently I have a DialPlan that filters calls and if accepted then dialplan simply Returns so that FreePBX Inbound Call Processing sends the call to Ring Group 600 (all phones) and that times-out after 30 seconds and send the call to 700 (Voicemail).
I now want to catch the timeout at 25 seconds so that I can send to Voicemail AND send a SMS. I looked at the DIAL command as this has a timeout but it only appears to work with Channels and non numeric extensions.
same => n,Dial(SIP/600, 25) gives ERROR numeric extension/ring group not allowed.
I’m not sure if SIP is applicable to Ring Groups. The extensions are CHAN_SIP and I have another Asterisk under development that will use CHAN_PJSIP.
This going to be a FreePBX error, but you’ve provided and Asterisk dialplan entry. How did you integrate the latter into the system? Maybe you are trying to override the FreePBX dialplan in an unsupported way? Ring groups are FreePBX concepts, so Asterisk would have no concept of them when presented with that dialplan line, not even the possibility that something called a ring group might exist.
Also you are using versions of the software which haven’t been supported for several years, although I don’t think that is the fundamental problem, here.
If 600 is a FreePBX ring group number, you probably need to dial it as something like Local/600@from-internal.
The error was in the Asterisk log. The DialPlan was the extensions_custom.conf file edited within FreePBX. I have not made any changes to any Asterisk file.
Ring groups are FreePBX concepts, so Asterisk would have no concept of them when presented with that dialplan line,
I didn’t know that!
I tried:
same => n,Dial(local/600@from-internal, 25)
and it works despite that in FreePBX only extensions have “from-internal” defined as a context (i.e. not Ring Groups).
Log shows:
Dial(“SIP/OBiTrunkSP2-0000001f”, “local/600@from-internal, 25”) in new stack
I get the SMS message and an email with the voicemail as an attachment
Thanks David for helping.
Alan
PS
Also you are using versions of the software which haven’t been supported for several years,
That’s why I have another system under development but it’s been a struggle to build a system from scratch (as a prebuilt Raspberry Pi/Debian system was no longer available) and to get a digital trunk working with a new VoIP telephony supplier (I currently interfacing to analogue line via OBi110). This new system went on hold but I need to resurrect soon before the analogue line gets switched off).
