On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN?
S
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Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, April 28, 2010 12:57:54 AM
Subject: Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, April 28, 2010 12:57:54 AM
Subject: Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN
Try: exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20) ?
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Thanks a lot Jim and Ryan.
It worked with changing the order as you suggested.
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Few more questions on Dial plan:
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media
also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding
On Tue, Apr 27, 2010 at 8:48 PM, Aditya Kumar adityakumar...@yahoo.comwrote:
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this done
I am not sure what your problem is. You can have a numeric extension dial an
alphabetic sip user.
exten = 123,1,Dial(SIP/somename)
The soft phone registers to your box with whatever username you set up.
If your phone can dial alpha then you can have
exten = alpha,1,Dial(SIP/$(EXTEN})
--
u pl give me complete
numbering plam
From: Jim Dickenson dicken...@cfmc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, April 27, 2010 7:09:45 PM
Subject: Re: [asterisk-users] Dial plan question.
I am
, April 27, 2010 10:11:16 PM
Subject: Re: [asterisk-users] Dial plan question.
Thanks a lot jim for the reply.
My issue is :
there is no numbers involved. I have soft clients.
when a user (bob) calls Alex,
he just opens his sip client and types in a...@pbx.com
so how can I write the translations
Hi All,
In our small office calls to the PSTN are currently sent via Asterisk and a
Linksys SPA3102 (1 x FXO and 1 x FXS):
SIP Phone -- Asterisk -- Linksys SPA3102 -- PSTN
If the PSTN is in use on SPA3102 I need a way to get the call to then route
out over IAX termination.
On Mon, 2007-07-30 at 16:09 +0100, Chris Blunt wrote:
If the PSTN is in use on SPA3102 I need a way to get the call to then
route out over IAX termination.
Usually, the best way to accomplish this is to send a call to your
Linksys ATA by using the Dial application from the dialplan, and then
Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
I am trying to do something that I see describe in a book and it is not
working
In my sip.conf, I have in my [fxo] context=from-pstn
I then have in extensions.conf
[from-pstn]
exten s,1,answer()
exten
Thanks, that set off a light bulb In my spa3K my incoming dialplan was
set to (S0:405)
Since this is a one FXO unit and my [from-pstn] will always be that line
can I make it generic and use the 's' extension as I described? If so what
would that spa3k dialplan be? just s0 ?
Doug
On Tue, 7
Answering my own question. If you want to connect an spa3K with
generic pstn inbound do the following...
for the pstn to voip dialplan in the pstn tab - (S0:ip-address-of-*)
in sip.conf
[sipurafxo]
context=from-pstn
etc.
Then in * extensions.conf use the s extension.
[from-pstn]
I am trying to do something that I see describe in a book and it is not
working
In my sip.conf, I have in my [fxo] context=from-pstn
I then have in extensions.conf
[from-pstn]
exten s,1,answer()
exten s,2,playback(blah)
etc.
It never answers but if I do this
[from-pstn]
exten
Hi List, this is probably quite straightforward
I need to call a sip extension for 15 seconds, if unanswered
I then need to call the same sip extension and an additional sip extension for
a further 15 seconds, finally if the calls arent answered I need it to
go to a generic unavailable
I need to call a sip extension for 15 seconds, if
unanswered I then need to
call the same sip extension and an additional sip
extension for a further 15
seconds, finally if the calls aren't answered I need
it to go to a generic
unavailable VM.
My question is if the first sip extension is
Chris:
One issue you might find, depending on the SIP phone at 4902, is that
it will show a missed call for the first 15 second attempt. If 4902
answers in the second 15 second attempt, it will still show a missed
call, when the incoming call was actually answered. If extension 4903
answers the
Hey everyone,
Hopefully this will be simple enough to answer. I have a menu setup like
below:
exten = 850,n,Set(MenuLoop=1)
exten = 850,n,Playback(mercury-prompts/welcome)
exten =
850,n(MainMenu),Background(mercury-prompts/MainMenu-if-you-know-the-ext)
exten =
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
says:
! wildcard, matches zero
or more characters immediately
(only Asterisk 1.2 and later, see note)
Note: The exclamation mark wildcard, which is
available only in Asterisk 1.2 and later, behaves
Try using . instead of !
_001800NXX
_X.
_X. is more like a match the rest instead of match all
Hope this helps.
Andy
On 3/22/06, Mike Hammett [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
says:
! wildcard,
Hi,
I have a basic dial plan question;
Here is the scenario.
Call comes through IAX and my * authenticate, then collect the digits and
dials out, simple :).
Here is the dial plan;
[did-in]
;for did callers
exten = 866219,1,Ringing
exten = 866219,2,Wait,4
exten = 866219,3,Answer
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