Hello,
I'm working on some dialplan rules to pull multiple users into a
conference call. I have some fairly straightforward rules which start
up a new MeetMe conference, allow escape with the * key to invite more
users, then use a features.conf sequence to bring the new user into
the conference w
http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
http://wiki.snom.com/Settings/overlap_dialing
Hope this helps - John
On Sun, 2009-07-26 at 05:07 +0100, hadi motamedi wrote:
> Dear Leif
> Can you please provide us with mo
Dear Leif
Can you please provide us with more details on this Overlap Dialing
phillosophy ?
Regards
H.Motamedi
On Wed, Jul 22, 2009 at 1:15 PM, Leif Madsen
wrote:
>
>
> John Novack wrote:
> >> Can you please let us know how we can modify our Asterisk
> >> "extensions.conf" file so it interprets
Dear John
The peer switch is Huawei switch and we need this functionality as we
support ISDN PRI but it supports for V5 interface . So another softswitch is
functioning between us . The peer side expects to receive the subscriber
dialed digits one-by-one as he sees us as an access equipment (not ju
I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does
anyone have a setup file they can share to help me work this out.
Instructions or a link I can follow - thanks.
___
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Elliot Murdock schrieb:
> Regarding using the sip show peers
> command, I remember somewhere seeing that it only works for static sip
> accounts and does not list accounts that are dynamically stored in a
> database. Most of my accounts are database entries, so would the sip
> show peers command w
Hello!
Thank you for the information. Regarding using the sip show peers
command, I remember somewhere seeing that it only works for static sip
accounts and does not list accounts that are dynamically stored in a
database. Most of my accounts are database entries, so would the sip
show peers com
Scott Gifford schrieb:
> Can you recommend a good tutorial or book that covers AEL?
http://www.das-asterisk-buch.de/2.1/extensions.ael.html has some
examples but unfortunately the explanations are in German. :-)
voip-info has some examples as well:
http://www.voip-info.org/wiki/view/Asterisk+AEL
On Saturday 25 July 2009 10:53:48 Scott Gifford wrote:
> Philipp Kempgen writes:
> > Miguel Molina schrieb:
> >> Philipp Kempgen escribió:
> >>> Use macros in AEL so you don't have to care about the underlying
> >>> implementation. :-) scnr
> >>
> >> Right now for every implementation I made, I di
Philipp Kempgen writes:
> Miguel Molina schrieb:
>> Philipp Kempgen escribió:
>
>>> Use macros in AEL so you don't have to care about the underlying
>>> implementation. :-) scnr
>
>> Right now for every implementation I made, I didn't have the need to
>> program in AEL, only plain extensions, so
2009/7/24 Louis-David Mitterrand :
> This used to work fine in 1.4:
>
> exten => 2131/,1,NoOp(reject3: ${CALLERID(num)})
> exten => 2131/,n,Playback(no_unknow_callerid_here)
> exten => 2131/,n,Hangup
>
> And now, after upgrading to 1.6.1.x it matches every callerid.
I'm not su
On Fri, 24 Jul 2009 21:27:12 -0400, David Cook wrote:
>I have installed them on a Linksys WRT54GL or WRT54GS v4/v3/v2/v1.1 devices.
>
>My mother-in-law's runs fine and she doesn't notice the difference. I know
>that is very subjective but to be honest I never looked at it for more than
>home-use/1
Hi All,
I have working asterisk 1.4.24.1, but I have issues with DeadAgi
application.
I am using hylafax and iaxmodem with asterisk, mail 2 fax and fax 2 mail
feature.
My system details are below:
OS: Centos 5.3
Asterisk Version: 1.4.24.1
Dahdi version: dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2
Za
Leif Madsen schrieb:
> I don't see how using the
> exten[/callerid] notation is really better than the GotoIf()
>
> Personally, the GotoIf() makes much more sense to me, because you're placing
> the
> matching logic in a single place,
True.
> as opposed to an error prone method of adding
>
Philipp Kempgen wrote:
> Louis-David Mitterrand schrieb:
>> On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
>>> On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten => 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten =>
Thanks Steve. But I couldn't find anything about a CSTA to AMI gateway for
asterisk at the quintum site. All I be able to find are TDM and FXO/FXS to
SIP gateways among others. I'm talking about (and I think gergis.rasmy too)
3rd party call control gateways, not interoperable gateways. By the way,
Miguel Molina schrieb:
> Philipp Kempgen escribió:
>> Use macros in AEL so you don't have to care about the underlying
>> implementation. :-) scnr
> Right now for every implementation I made, I didn't have the need to
> program in AEL, only plain extensions, some AMI and AGI. But well, it
> see
On 25/07/09 00:08, John A. Sullivan III wrote:
> Hello, all. After many pages of googling and testing in the lab, I'm
> still a bit perplexed about how to implement tls protection for the
> asterisk manager. manager.conf allows one to specify the cert file but
> one normally must also specify the
Hi,
I'd like to disable MWI on certain lines of my IP650 Polycom phone. So I
removed the "mailbox=" parameter from that line's peer section in
sip.conf. Yet the envelope still appears in front of that line and the
phone MWI keeps blinking.
Where should I look to completely disable MWI on a certai
On Fri, Jul 24, 2009 at 11:14:47AM +0200, Philipp Kempgen wrote:
> Louis-David Mitterrand schrieb:
> > On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
> >> On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
> >> >
> >> > This used to work fine in 1.4:
> >> >
> >> > exten =>
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