Just done it ... and all works fine.
Thanks all.
Marco
2009/7/24 Administrator TOOTAI ad...@tootai.net
Marco Sambo a écrit :
Hi all,
I've a problem: I update my asterisk to version 1.4.25, and the attended
transfer doesn't work.
[...]
Marco,
attented transfer are broken in
2009/7/22 gergis.rasmy gergis.ra...@gmail.com
does Asterisk suppoet CSTA protocol for CTI applications?
No it doesn't but I've heard some gateways exist (software translating CST
to AMI).
Regards
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2009/7/21 Loic Didelot ldide...@mixvoip.com
Hello,
I played with the externalIVR application. So far I am able to read
digits and play sound files. But how can I leave the application and
continue in the dialplan so that I can execute other actions like going
to voicemail, or ringing users
Hi,
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.
Did something change?
Thanks,
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
Hi,
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
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 = 2131/,n,Playback(no_unknow_callerid_here)
exten =
Heelo,
I currently search a program that can make a web browser Pop-up on an
incoming call on a specific URL like :
http://directorie.ch?CALLNUMBER:00451849799
I have found ADM, but it's a bit more complex for my purpose an it's not
very stable.
Do you know a simple software for that ?
An
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: ${CALLERID(num)})
exten =
Not FAX over VoIP, but testing 2 FAX machines back to back ...
The scenario is that a client has one of my asterisk VoIP only systems
which they're happy with, but need to test FAX machines, so rather than
plug in a TDM card with FXS ports, I'm suggesting something like a PAP2T
device with 2
On Fri, 24 Jul 2009, Vincent Renaville wrote:
Heelo,
I currently search a program that can make a web browser Pop-up on an
incoming call on a specific URL like :
http://directorie.ch?CALLNUMBER:00451849799
I have found ADM, but it's a bit more complex for my purpose an it's not
very
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Remember, setvar in sip.conf makes that variable a global variable.
Andrew Thomas
Technical Services Manager
Juan C. Crespo R.
Vincent Renaville schrieb:
An other part of my project is to eneable click-to-call from a web page, do
you know a kind of project that implement callto protocol, at this time I
use Noojee click but It only work with Firefox.
BTW: Sooner or later every major web browser will implement custom
Hello,
Did anyone succeeded in installing Asterisk on OpenWRT system.
pls help.
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2009/7/24 Philipp Kempgen philipp.kemp...@amooma.de
Vincent Renaville schrieb:
An other part of my project is to eneable click-to-call from a web page,
do
you know a kind of project that implement callto protocol, at this time I
use Noojee click but It only work with Firefox.
BTW:
Hi everybody
In advance sorry for my bad english and if my problem was already exposed (I
didn't find any tips in the mailing list archive. Bad luck)
I have some questions about asterisk 1.6 release :
1) how can I do a n+101 priority jumping if a SIP canal is busy ?
I read that the general
harry R wrote:
Hi everybody
1) how can I do a n+101 priority jumping if a SIP canal is busy ?
I read that the general parameter priorityjumping is depreciated in
the 1.6 release and I already try the j option in dial() application
but no way.
You'll want to use DIALSTATUS
exten =
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Juan C. Crespo R. wrote:
Guys I wonder if its possible to set a different MoH based on
Here's how I think your dialplan should look:
exten = 101,1,Ringing
exten = 101,2,Answer()
exten = 101,3,Dial(SIP/quentin,10)
exten = 101,n,VoiceMail(1...@default,u)
exten = 101,n,Playback(vm-goodbye)
exten = 101,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
exten = 101-BUSY,n,Wait(3)
Hello,
i´ve a question about the Meetme Options. How could i play a enter and
leave sound but without recording the user name first. I just want a
User joined conferenc and a user leaved.
With the i or I Option i have to record the name first.
Is there any way of doing this? As i can see in the
Here's my solution to the OP's question
- exten = s,1,answer
- exten = s,2,Set(MOH=$DB(OHCLASS/${EXTEN}))
- exten = s,3,SetMusiconhold(${MOH})
- exten = s,4,Dial(SIP/${EXTEN}),30,ikKtTm)
Just set OHCLASS exten to the value you want in advance
-Original Message-
From:
harry R schrieb:
2) about asterisk voicemail maximum message limit, is it possible to send a
notification mail to an user if his vmbox is full ? How can i do that if
it's possible.
Write a cron job to check if one of the mailboxes is full
(ls -l
Yeah I try Doug solution and It works !
Thanks everybody
2009/7/24 Danny Nicholas da...@debsinc.com
Here’s how I think your dialplan should look:
exten = 101,1,Ringing
exten = 101,2,Answer()
exten = 101,3,Dial(SIP/quentin,10)
exten = 101,n,VoiceMail(1...@default,u)
exten =
Hi
I have a new question. Here the situation :
I use softphone on 2 computers (soft1 and soft2) located on the same
subnetwork.
When I register on asterisk server using soft1 with one user (e.g JOHN)
which I declared in sip.conf I can register again with this same user using
soft2.
Is it normal ?
Set John's defaultip to 192.168.1.1 in sip.conf
Defaultip=192.168.1.1
This should tie john to that IP address.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of harry R
Sent: Friday, July 24, 2009 9:23 AM
To:
On Fri, Jul 24, 2009 at 9:22 AM, harry Rrhm.noa...@gmail.com wrote:
How can I say to asterisk server : don't accept other registration for
JOHN if you see that he's already registred on another softphone ?
In sip.conf you can set permit and deny statements based on IP address.
[JOHN]
Yeah, have it running on several units. It's really quite simple now.
- Goto System - Packages
- Scroll down to Update Package List and wait a few seconds for that puppy
to refresh.
- You now should have a list of installed packages followed by a very long
list of available packages.
- Find the
Gordon Henderson wrote:
Not FAX over VoIP, but testing 2 FAX machines back to back ...
The scenario is that a client has one of my asterisk VoIP only systems
which they're happy with, but need to test FAX machines, so rather than
plug in a TDM card with FXS ports, I'm suggesting something
We'll soon be releasing a second beta. Contact me off list for more
information.
It will do both screen pop and enable dialing with Internet Explorer.
Regards,
Elliot
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent
Does anyone have a list of Skype or other Anonymous VOIP end point phone
numbers that I can use to block unsolicited calls with?
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What router did you install it on? Any stats on concurrent conversations
/ transcoding?
Cheers,
j
On Fri, 24 Jul 2009, David Cook wrote:
Yeah, have it running on several units. It's really quite simple now.
- Goto System - Packages
- Scroll down to Update Package List and wait a few
Do you know what names those gateways have?
Jose
2009/7/24 Olivier oza-4...@myamail.com
2009/7/22 gergis.rasmy gergis.ra...@gmail.com
does Asterisk suppoet CSTA protocol for CTI applications?
No it doesn't but I've heard some gateways exist (software translating CST
to AMI).
Regards
I was going to suggest a custom browser/jabber plug in, as I have done in
the past, but a browser window had to be open for it to work.
Looks like time has caught up with need.
I will be checking this out.
Thanks,
Steve T
On Fri, Jul 24, 2009 at 11:37 AM, Elliot Otchet
It also installed just fine on a very flaky Dell router that came free with
an order.
Check out the OpenWRT site, they have a matrix of compatible routers.
On Fri, Jul 24, 2009 at 1:54 PM, Steve Totaro
stot...@first-notification.com wrote:
I have played with it a while ago on a WRT54GS, the
I have played with it a while ago on a WRT54GS, the old style, pre vers 4 or
whenever they changed to two models, one running VxWorx and the other
specified for Linux Enthusiasts.
It is best to get an old one. I am not positive that what I said above is
totally correct as far as version or OS
Without any research, I would check out the Quintum lineup.
They have feature sets which are amazing (and confusing as all heck) and
work great once configured.
Thanks,
Steve T
On Fri, Jul 24, 2009 at 1:03 PM, Jose Arias cyr2...@gmail.com wrote:
Do you know what names those gateways have?
My skype number appears to belong to a pool given to Level 3
Communications, and it was out of the same 1000 block as my Google Voice
number. You could block all Level 3 numbers for your area, but it would
run the risk of blocking legitimate customers from calling you.
Casey Boone
Jared
harry R schrieb:
2) about asterisk voicemail maximum message limit, is it possible to
send a notification mail to an user if his vmbox is full ? How can i do
that if it's possible.
On Fri, 24 Jul 2009, Danny Nicholas wrote:
for question 2, this depends on how voicemail.conf is setup. By
The Asterisk Development Team has announced several Asterisk-Addons releases,
including Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1. These releases are
available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
These releases are an incremental release after some
Anybody seen this article yet? Looks like Russian Telecom business have
decided that VoIP is going to put a dent in their profits so their
pitching it as a threat to Russia's national security and working to get
laws put into place to make sure the government controls VoIP providers
operating
Good luck with that.
Brent Davidson wrote:
Anybody seen this article yet? Looks like Russian Telecom business have
decided that VoIP is going to put a dent in their profits so their
pitching it as a threat to Russia's national security and working to get
laws put into place to make sure
Hello,
I'm trying to implement multi-party calls according to these
instructions:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
They are almost working, except that the Goto at the end of
[dynamic-nway-start] doesn't seem to work. When I turn verbosity up a
bit, I get
Alex Balashov wrote:
Good luck with that.
Brent Davidson wrote:
Anybody seen this article yet? Looks like Russian Telecom business have
decided that VoIP is going to put a dent in their profits so their
pitching it as a threat to Russia's national security and working to get
laws put
Try replacing the , with |. Shouldn't make any difference, but it is a
future deprecation if I remember correctly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Gifford
Sent: Friday, July 24, 2009 3:14
Scott Gifford escribió:
Hello,
I'm trying to implement multi-party calls according to these
instructions:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
They are almost working, except that the Goto at the end of
[dynamic-nway-start] doesn't seem to work. When I turn
Miguel Molina schrieb:
I just ran into a similar problem, I needed a macro spreaded over
several contexts because it's kind of a part of an IVR. I switched to
GoSob() and Return() applications
(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub) and
everything goes fine now.
I'm working on a script that needs to determine the extension (eg: 123) of
the phone that initiated the call, or CALLERID number if an externall
caller.
Is there a simple way to do this?
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${CALLERID(num)} should provide this information in both cases.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, July 24, 2009 4:43 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] How
Michelle Dupuis schrieb:
I'm working on a script that needs to determine the extension (eg: 123) of
the phone that initiated the call, or CALLERID number if an externall
caller.
Is there a simple way to do this?
${CALLERID(num)} ?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 -
On Mon, May 4, 2009 at 2:36 PM, Jeff LaCoursierej...@jeff.net wrote:
On Mon, 4 May 2009, Jimmy Godbout wrote:
Take a look at wrp400 from Linksys/Cisco. It has 2 fxs, 802.11g and 4
switched ports + wan.
Sigh, I suppose this is exactly what I was talking about :) You guys sure
know how to
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 line applications. Can't say I've had a problem that
The way that I understood this to work was that e164.org lists all toll-free
numbers to make it free to call those kinds of numbers (instead of using one
of your own trunks). Since ENUM can provide priorities, if I own a toll-free
and enter it into the system, the route that I specify will be
Does anyone have an example of how to create a custom filename for the
(combined in/out) audio file captured through automon?
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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 private key file. If I simply enter
the cert file:
ENUM lookups at e164.org return a IP route for ALL toll-free numbers.
I was surprised to observe that ALL toll-free numbers get a hit at e164.org.
It appears that ALL toll-free prefixes have been delegated, thereby
publishing an IP route for YOUR TOLL-FREE NUMBERS, my toll-free numbers, and
Philipp Kempgen escribió:
Miguel Molina schrieb:
I just ran into a similar problem, I needed a macro spreaded over
several contexts because it's kind of a part of an IVR. I switched to
GoSob() and Return() applications
(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub) and
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