I live in Azerbaijan. There are city phone numbers like *8200, *6000. How can i configure outbound routes in freepbx or asterisk configuration files to call numbers with * symbol? Please, help me. It is very important for me.
Please, help me.
Please, help me. I am waiting for your answer yet.
Try with extensiones_customs.conf
exten => 4444.,1,Dial(DAHDI/g0/{EXTEN:4},20)
Latre your solution makes no sense?
How does an extension for 4444 that dials the contents of the current extension starting at the 4th position going to help.
I just tested with FreePBX 2.11, and outbound routes has no problem including a * character in dial patterns. As long as you have an outbound route defined with a match pattern of “*XXXX” without the quotes, it should work. I can’t test with older version of FreePBX.
Thank you lgaetz. I tested this pattern with freepbx 2.10 and it did not works. On monday i will try it with freepbx 2.11.
As far as I know it worked even on the earlier versions. I used that * symbol on the dial plan.
johnny2000 , please , write down your dial plan with * symbol and i will test it.
One thing that you did not mentioned yet, are you using a SIP trunk for your outbound calls? Can you post your trunk configuration?
4444 is only example…i tought that you know about dialplan…
Its 4444. because when yo dial 4444*XXXX dial but :4 remove first 4 numbers, so only dial *XXXX…and go for trunk dahdi g0.
If you dont know what it is this…dont do it.
My SIP trunk configuration:
Trunk name - 701
Outbound caller id - 5399999 (our did number)
Peer details:
type=friend
secret=xxx
host=dynamic
dtmfmode=inband
context=from-trunk
I am using Grandstream FXO GXW 4104.
I just tested this pattern with freepbx 2.11 , but it does not working.
When i dialing *XXXX number, in asterisk cli i show this warning:
WARNING[1926][C-00000005]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
Asterisk 11.6, freepbx 2.11
There is something not right with your host, it is supposed to be your provider that terminates the outbound calls from your PBX. Basically based on your Trunk Configuration, you are not terminating the call.
My problem with dialplan, outbound routes is not solved yet. Please, help me.
your trunk definitions look very odd. check with your provider for the proper trunk definitions. you need to have a host name at least.
my trunk config is compatible with fxo gxw 4104. my asterisk not connected to any sip provider, we are using analogue pstn provider. therefore i dont need hostname in trunk config. Our all calls to pstn is working, else to *XXXX pstn numbers.
Then you should have a DAHDI/ZAP trunk configuration. Can you post it?
Trunk configuration
My SIP trunk configuration:
Trunk name - 701
Outbound caller id - 5399999 (our did number)
Peer details:
type=friend
secret=xxx
host=dynamic
dtmfmode=inband
context=from-trunk
I am using Grandstream FXO GXW 4104.