On Fri, 2011-06-10 at 05:52 -0400, Steve Totaro wrote:
> On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet wrote:
> > On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
> >> On Thu, 9 Jun 2011, Hans Witvliet wrote:
> >>
> >> > I went originally from a almost working machine running:
> >> > aster
Queue not sending call to Agent
I am having an issue and i am not sure if it is a bug or a config issue. I
was originally running Asterisk 1.8.1.1 when I noticed this issue. I
upgraded to 1.8.4.2 to see if that would fix it but it didn't.
The issue is that I have a call queue and the agen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall
Sent: Friday, June 10, 2011 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Recording
Longtime lurker, first ti
Longtime lurker, first time poster. :)
A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route. I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.
reco
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Jerry Geis
> Sent: Friday, June 10, 2011 2:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] AMI questio
Sure,
Start call
[Jun 10 10:01:17] VERBOSE[27776] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 10 10:01:17] VERBOSE[7269] pbx.c: -- Executing [8004815122@from-sip:1]
Dial("SIP/7081-05ed", "DAHDI/g1/18004815122") in new stack
[Jun 10 10:01:17] DEBUG[7269] sig_pri.c: sig_pri_request 5
Howdy,
I am playing around with asterisk within an LXC container on Ubuntu
11.04. I have asterisk (1.4.42) running fine, but want access to
dahdi_dummy for timing (meetme). I have dahdi installed on the "host",
and dahdi_dummy is loaded:
root at astnorth:/# ls -ltr /dev/dahdi
total 0
crw-rw-
Either use ExtensionState or watch for Hold/Unhold events.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState
Eric
Thanks I have 2 questions:
1) trying to use the command and not getting anything of value:
Response: Success
ActionID: 1
Message: Extension Status
On Fri, Jun 10, 2011 at 1:27 PM, satish patel wrote:
> Hi,
>
> We having some PRI call drop issue on asterisk 1.8.x but we had no issue
> never ever on asterisk 1.2. Anybody else having this issue ?
>
> -S
>
>
What troubleshooting have you already done? Do you have logs? Debug
information? Wh
On Fri, Jun 10, 2011 at 9:48 AM, Jamie A. Stapleton <
jstaple...@computer-business.com> wrote:
> Many providers do not allow for caller ID name to be sent.
>
>
In the US, there are CNAM databases that not all providers have access too.
For example, flowroute, even though they are an excellent VOIP
On Fri, Jun 10, 2011 at 12:52 PM, Warren Selby wrote:
> I'm on my phone, otherwise I'd give example dialplan and sip.conf snippets.
>
>
I'm back home, so here's some examples. I'm using a template in sip.conf
that provides a lot of the common, duplicated settings for the phones on
site, like typ
I have expanded the EWS calendar functionality within Asterisk 1.8 so it
is now possible to access any calendar within an Exchange 2007 or 2010
server.
I have put the changes onto the reviewboard for astrisk but currently no
one responded.
So if you use the EWS calendar functionality within As
Hi,
We having some PRI call drop issue on asterisk 1.8.x but we had no issue never
ever on asterisk 1.2. Anybody else having this issue ?
-S
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What I do is add a setvar=extclid=xxx to each entry in sip.conf, matching the
appropriate external callerid with its respective internal extension. Then, in
my outbound context, just before dial, I set CALLERID(num)=${extclid}.
I'm on my phone, otherwise I'd give example dialplan and sip.conf s
On Fri, 2011-06-10 at 16:31 +0530, virendra bhati wrote:
> Hi List,
>
> I don't install from yum repository. I download tar file from
> asterisk.org
>
>
>
> On Fri, Jun 10, 2011 at 3:03 PM, Alexandru Oniciuc
> wrote:
> What do you mean?
>
> Did you installed from sour
Le 10/06/2011 08:07, Florent THOMAS a écrit :
Hy,
Does anybody knows how to show the digium addons in the freepbx GUI.
The module is available in the GUI but sadly empty!
Everything seems to be correctly installed bute the tables in the
database are totally empty.
Is there any script anywhere
Many providers do not allow for caller ID name to be sent.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Friday, June 10, 2011 5:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-
Either use ExtensionState or watch for Hold/Unhold events.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Jerry Geis
> Sen
Good morning gentlemen, is my first post in the list, now I'm starting asterisk
wanted to have your help for some questions.
Well the first function is as follow me. Here
I will demonstrate how this configuration follow me on my
extensions.conf but it is not working, and do not know why, but
Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X
Thanks,
jerry
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New to Asterisk? Join us for a live introductory webinar ever
On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles
wrote:
> On Friday 10 Jun 2011, Steve Totaro wrote:
> > I never understood hy people who have block of DIDs in a row choose to
> > make life difficult by not incrementing extensions by one, send caller
> > ID by prepending the common numbers and only sen
On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles
wrote:
> On Friday 10 Jun 2011, Steve Totaro wrote:
> > I never understood hy people who have block of DIDs in a row choose to
> > make life difficult by not incrementing extensions by one, send caller
> > ID by prepending the common numbers and only sen
Arjan Kroon | Mobillion wrote:
But are there also pathes for version 1.6
The last patch available for the 1.6 series was for 1.6.0.1:
https://issues.asterisk.org/jira/browse/8824
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
On Friday 10 Jun 2011, Steve Totaro wrote:
> I never understood hy people who have block of DIDs in a row choose to
> make life difficult by not incrementing extensions by one, send caller
> ID by prepending the common numbers and only sending four digits.
Well, to be fair, that's what most people
On Fri, Jun 10, 2011 at 4:00 PM, Steve Totaro
wrote:
> On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta
> wrote:
> > Hi,
> > I have 44578900 to 44578999 DID's. and I have extensions(100) for this
> > DID's. but problem is
> > callerid Extensions
> > 44578900 100
> > 44578901 101
-Original Message-
From: Steve Totaro
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 10 Jun 2011 06:30:53
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call
We have two systems one with version 1.6 and one with version 1.8
With 1.8 we don't see the problem
Unfortunately it is not possible to upgrade 1.6 to 1.8.
But are there also pathes for version 1.6
Arjan Kroon
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mail
On Friday 10 Jun 2011, Steve Totaro wrote:
> Why do programmers try to make solution so elegant when an entries for
> each phone in sip.conf is all that is needed.
>
> No need for mathematical formulas, AGIs, and databases. You just took
> over engineering to a new level.
Because doing it your wa
Steve Totaro wrote:
For each phone, add "callerid="Joe Smith"<1551212> " no quotes in sip.conf
The problem with that solution is that station to station calls will
show the same CID and not the extension.
I'd vote for the database approach.
Doug
--
Ben Franklin quote:
"Those who wo
Hi List,
I don't install from yum repository. I download tar file from asterisk.org
On Fri, Jun 10, 2011 at 3:03 PM, Alexandru Oniciuc <
alexandru.onic...@trivenet.it> wrote:
> What do you mean?
>
> Did you installed from sources or distro packet?
>
>
>
> sources: make uninstall
>
> distro: Ev
Arjan Kroon | Mobillion wrote:
Does anybody have problems with a wrong Connected Line ID with asterisk version
1.6
As far as I know, unless you're applying patches yourself, Connected
Line ID is only available for the 1.8 series. I'm running it on 1.4
with patches.
Doug
--
Ben Franklin
Hai,
Does anybody have problems with a wrong Connected Line ID with asterisk version
1.6
The following bug was for version 1.4, but I cannot make up if this bug is
still in version 1.6
http://forums.digium.com/viewtopic.php?t=7780
In version 1.8 it is possible to change the Connected Line ID,
On Fri, Jun 10, 2011 at 6:27 AM, A J Stiles
wrote:
> On Friday 10 Jun 2011, mahesh katta wrote:
>> Hi,
>> I have 44578900 to 44578999 DID's. and I have extensions(100) for this
>> DID's. but problem is
>> callerid Extensions
>> 44578900 100
>> 44578901 101
>> 44578902 102
On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta wrote:
> Hi,
> I have 44578900 to 44578999 DID's. and I have extensions(100) for this
> DID's. but problem is
> callerid Extensions
> 44578900 100
> 44578901 101
> 44578902 102
> 44578902 103
> 44578903 104
> 44578905
On Friday 10 Jun 2011, mahesh katta wrote:
> Hi,
> I have 44578900 to 44578999 DID's. and I have extensions(100) for this
> DID's. but problem is
> callerid Extensions
> 44578900 100
> 44578901 101
> 44578902 102
> 44578902 103
> 44578903 104
> 44578905 200
>
On Fri, Jun 10, 2011 at 1:48 AM, Hans Witvliet wrote:
> On Thu, 2011-06-09 at 16:32 -0700, Steve Edwards wrote:
>> On Thu, 9 Jun 2011, Hans Witvliet wrote:
>>
>> > I went originally from a almost working machine running:
>> > asterisk180-1.8.3.2-87.1
>> >
>> > To a machine that continuously restar
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906 275
44578907 277
44578908 354
I need to
On 2011-06-10 07:30, virendra bhati wrote:
Hi John,
Sorry for wrong information. Actually it's J not P option in
ControlPlayBack...
http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback
That page is correct for asterisk 1.4 but the feature you need is in
1.6.0 and forward.
Have yo
What do you mean?
Did you installed from sources or distro packet?
sources: make uninstall
distro: Every distro has its own commands (yum, apt-get ecc)
Alex
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di virendra bhati
Inviato: venerdì
Hi List,
I want to set my caller ID and name with asterisk. So that when I make
outgoing calls then destination end will see my name with number.
from asterisk end I set both the things into dialplan.
---
--
exten => _X.,n,Set(CALLERID(num)=9172341457)
exten => _X.,n,Set(C
Maybe you could think about a root command line like that :
yum uninstall asterisk
Le 10/06/2011 11:26, virendra bhati a écrit :
Hi List,
Is there any way by which we can remove asterisk from machine without
deleting folder manually? I did google and gets various solution by no
success. even
Hi List,
Is there any way by which we can remove asterisk from machine without
deleting folder manually? I did google and gets various solution by no
success. even after deleted asterisk will be there .
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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