Discussion: Asterisk config extensions.conf

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yes man, it is very good to use. Using it for months now and never had issues with it!
Cheers from Tom @ Beste Wettanbieter

by tooom, Saturday 14 of May, 2011 (22:28:51 UTC)
Press-Star for re-enter

I need to add a line which will let callers to enter * if they have made a mistake entering the extension.

for maple if they have entered 123 and they want to clear it all and enter 1245

by amirsp55, Wednesday 28 of May, 2008 (22:39:01 UTC)
Re: Extentions

It's pretty simple to manipulate sip extensions with Asterisk. I see no one answered this so here's the soluion...

; --- user dials 9+ and we manipulate dialled digits ---

; --- just wait two seconds before dialling the extension ---
exten => _9.,1,Wait(2)

; --- prefix + and strip off user dialled 9 then place call for two seconds ---
exten => _9.,n,Dial(SIP/+${EXTEN:1}@sipcarrier,2)

; --- prefix + and strip off user dialled 9 then place call to othersipcarrier ---
exten => _9.,n,Dial(SIP/+${EXTEN:1}@othersipcarrier)

by jnfuller, Sunday 20 of May, 2007 (15:40:37 UTC)

I have been working on my asterisk exstention configurations and I would like to know the syntax for making asterisk prefix a number to a dialed exstention. Also I would like to know how to make asterisk wait 2 seconds before dialing an exstention. If possible I would also like to know the syntax to make asterisk dial a predetermined number, wait for two seconds, and then dial the extention.

by emawk05, Tuesday 05 of December, 2006 (16:16:44 UTC)
Works in 1.2

This works in 1.2 and later now.

by forrest.beck, Friday 28 of July, 2006 (15:57:17 UTC)

by RichardBG, Saturday 07 of January, 2006 (22:14:52 UTC)
What special symbols are allowed in dial plans?

This article mentioned that "the name of a extension can contain any letter or number as well as some punctuation marks." Which punctuation marks are supported? I gather using '.' is a bad idea, and I doubt ',' would work either, but what's safe?



by g, Saturday 26 of November, 2005 (22:14:12 UTC)

hi ive been setting up asterisk and have stumbled upon a problem that i cannot seem to solve or get any advice on now i initially used the asterisk@home AMP to set up my trunks and in there when i add a sip trunk is an option called 'MAximum Channels' which explains that this is the maximum amount of channels both incoming and outgoing that can be active for this trunk at any one time, however this is not the case if i specify 2 then i can only place 2 outgoing calls but can still receive as many calls as i wish which in turn kills my bandwidth, i would expect a caller to get an engaged tone if the maximum amount of channels was exceeded. Can i do this with dialplan??


by mattbrown7, Wednesday 05 of October, 2005 (19:13:49 UTC)

by ,