On Thu, 19 Aug 2010, Tino wrote:
But when i call my DID number following dialplans are being executed.
What i need is to set a variable with one value for one DID number and
set the same variable with another value for another DID number. Also
any contexts should be able to use this
On Thu, 2010-08-19 at 16:56 -0500, Carlos Chavez wrote:
I am making a web interface so users can manage their voicemail. The
only problem I have is that since the Web server and Asterisk run as
different users I need to run some commands through Asterisk so I can
manipulate the
Paddy,
I believe I have a solution, let me sober a bit ;) and rum it through
(typo not intended but funny) my test server to doublecheck
Sent from my iPhone
On Aug 20, 2010, at 12:20 AM, Paddy Grice pa...@wizaner.com wrote:
Hi Sherwood
I actually do want dynamic CLID as I tried to make
A boddingtons...
Anyway, let me point out that the CONTEXT has nothing to do with
'access' to a variable...if a call (channel) causes a variable to be
assigned a value, then that calland possibly it's 'children' if
inheritance is set up. It doesntmatter what context the call ends up
On Thu, 19 Aug 2010, Tino wrote:
But when i call my DID number following dialplans are being
executed. What i need is to set a variable with one value for one
DID number and set the same variable with another value for another
DID number. Also any contexts should be able to use this
Hey Steve, I'm not drunk. I was attempting to keep some levity in my
posts because I was pretty frickin irate at the time of several posts
that night.
Or, maybe I AM drunk, and I can still remember all sorts of nifty
Asterisk dialplan and related administration stuff while plastered
Make a html link this way:
a href=
http://widget.doddlephone.com/embed/webphone.jsp?sipserver=SERVERusername=USERpassword=PASSWORDcallto=PHONETOCALLauto=yes
Tel: +1 234 567 890
/a
/b
Sergio
On Fri, Jul 9, 2010 at 5:29 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
What would you suggest to get
Hi,
Thanks. Actually can it be done on whole kit basis rather than for an
extension or peer. Like if there are lot of inbound sip interconnects on a
kit , how can we send first 50% simultaneous calls to dahdi with codec A and
after that with codec B.
Thanks,
D
--
Hi,
i'm trying to get PoC on Nokia Phones to work with asterisk. I think the
store-and-forward part could easy be done in the dialplan, but i can't even
get the handset to register with asterisk (authentication failed). I'd try'd
to find the difference between pure sip and PoC-SIP, but didn't
On 19 Aug 2010, at 20:59, Randy R wrote:
On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com
wrote:
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
This island runs it's GSM network on OpenBTS: http://www.niueisland.com/
This was
2010/8/20 Doddle WebPhone doddleph...@gmail.com
Make a html link this way:
a href=
http://widget.doddlephone.com/embed/webphone.jsp?sipserver=SERVERusername=USERpassword=PASSWORDcallto=PHONETOCALLauto=yes
Tel: +1 234 567 890
/a
/b
Sergio
Hi,
Yes, adding this kind of link should do
On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote:
On 19 Aug 2010, at 20:59, Randy R wrote:
On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com
wrote:
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
This island
On Wednesday 11 Aug 2010, Tilghman Lesher wrote:
On Wednesday 11 August 2010 03:59:28 A J Stiles wrote:
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon.
With some calls, the value in the `billsec` field in the CDR is exceeding
the value in the `duration` field.
I'd
With all honor and respect you deserve, Do I need your permission to
express my point of view on community forum ?
also it would be quiet helpful for us if you understand well
the requirement of post
Regards
On Fri, Aug 20, 2010 at 1:34 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com
Hi Tim, I'm not a radio guy too!
I saw your name on the test in Niue.
I have a softswitch. Can I replace Asterisk by my softswitch?
Thanks
On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk
On Friday, August 20, 2010 10:35:10 am Olivier wrote:
Yes, adding this kind of link should do it but I'm looking for a solution
which automatically insert whatever is needed to launch a call.
wouldn't it be difficult to know exactly which applications are available on
the system which has the
Thanks for all the help, but I still can't find what's wrong.
I enabled console = notice,warning,error,debug,dtmf like Miguel suggested.
The output is attached.
I noticed that the rtp.c session never starts, which as I understand is what
catches the dtmf tone, but I could not find how to start
1. Set up a Global Variable that will store that kit's current number of calls
2. Check that variable when a call starts (but before you dial out)
3. If the number of calls is 49 (since the current call will make
50), use codec A via the CHANNEL() function, otherwise use codec B
using the same
Nasir Iqbal na...@ictinnovations.com wrote:
With all honor and respect you deserve, Do I need your permission to
express my point of view on community forum ?
also it would be quiet helpful for us if you understand well
the requirement of post
*snip*
Nasir,
You don't need my permission to
On Fri, 20 Aug 2010, Sherwood McGowan wrote:
1. Set up a Global Variable that will store that kit's current number of calls
2. Check that variable when a call starts (but before you dial out)
3. If the number of calls is 49 (since the current call will make
50), use codec A via the CHANNEL()
Steve,
Good point my man...You drinking yet? LOL...I had forgotten about the
GROUP and GROUP_COUNT functions, that is a much better way (in that it
already existed and doesn't require me to write more code :] )
Slainte!
On Fri, Aug 20, 2010 at 7:37 PM, Steve Edwards
asterisk@sedwards.com
On Fri, 20 Aug 2010, Sherwood McGowan wrote:
Good point my man...You drinking yet?
Let me check to see if I still have a pulse -- yep!
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice:
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