Jim Dickenson wrote:
> Building modules, stage 2.
> MODPOST
> WARNING: could not find
> /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.
> o.cmd for
> /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o
>
> Anyone else seeing this?
>
You
I download the tar.gz file and expand it, without error. I am not sure how I
could not have a complete download.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
> From: Tzafrir Cohen
> Organization: Xorcom*
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussi
On Wed, Jun 24, 2009 at 03:53:18PM -0700, Jim Dickenson wrote:
> I have an i686 cpu and when compiling from source I get this error:
>
> touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified
> Building modules, stage 2.
> MODPOST
> WARNING: could not find
> /usr/src/dahdi-
On Wed, Jun 24, 2009 at 05:41:34PM -0400, Steve Totaro wrote:
> On Wed, Jun 24, 2009 at 5:31 PM, Tzafrir Cohen
> wrote:
>
> > On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote:
> >
> > > In FreePBX there are whatever_custom.conf files that are not touched when
> > > changes are made in
I have an i686 cpu and when compiling from source I get this error:
touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified
Building modules, stage 2.
MODPOST
WARNING: could not find
/usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.
o.cmd for
/usr
I need to set the II digits for some outgoing calls originating with
asterisk, but the documentation seems to show that all the various
ANI2 variables are read-only. So how do I set them?
(Yes, we have Feature Group D trunks and allowed to set them and
regularly do with our C.O. switch. Th
On Wed, Jun 24, 2009 at 5:31 PM, Tzafrir Cohen wrote:
> On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote:
>
> > In FreePBX there are whatever_custom.conf files that are not touched when
> > changes are made in the GUI.
>
> The _custom file is not touched. But it is merely part of the
>
On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote:
> In FreePBX there are whatever_custom.conf files that are not touched when
> changes are made in the GUI.
The _custom file is not touched. But it is merely part of the
configuration file. And what if you want the luser to be able to
c
I think I got it. ${DIALEDPEERNUMBER} contains the leg that connected
(just what I need).
FYI I used DumpChan() to get all the available variables and found it.
Thanks!
Enlai
On Wed, 24 Jun 2009 14:19:46 -0700, asterisk-users@lists.digium.com
said:
> Thanks Danny.
>
> I tried accessing ${CHANNE
Looking at my man php5 q is not a valid option. That may be just on
Suse.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of "Juan E.
Rodríguez"
Sent: Wednesday, June 24, 2009 4:18 PM
To: Asterisk Users Mailing List - Non-C
The phone caches the configuration... To remove it update the config like
so:
line2_name:"UNPROVISIONED"
line2_authname:"UNPROVISIONED"
line2_password:"UNPROVISIONED"
line2_shortname: "UNPROVISIONED"
line2_displayname: "UNPROVISIONED"
For each line that you don't want anymore.
Try running your script with /usr/bin/php5 script.php to test it
Or changing #!/usr/bin/php5 -q to #!/usr/bin/php -q
Leah Newmark wrote:
Thanks.
I didn't change anything in my dialplan. I am aware of reloading configuration
:)
My AGIs are copied from a working asterisk install -- the sheba
Folks,
I have CISCO 7940g phone. I have in the past configured the phone with
two lines. Having found the 2nd line wasn't much use, I want to remove
it from the config. I have taken it out of the SIP config file that is
TFTPd to the phone but it is still showing on the phone and it is still
try
On Wed, Jun 24, 2009 at 4:39 PM, Tzafrir Cohen wrote:
> On Wed, Jun 24, 2009 at 09:20:44PM +0200, jonas kellens wrote:
> > I wonder if there is a GUI that does not change the underlying hand-made
> > configuration ?!
> >
> > What I'm looking for actually is a GUI for adding a new SIP-client +
> >
${CHANNEL} or ${DNID} should do the trick.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-us...@enlai.net
Sent: Wednesday, June 24, 2009 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discuss
On Thu, 2009-06-25 at 10:56 -0700, gmail wrote:
> i am trying to make a video call on asterisk 1.6
Video support in Asterisk 1.6.0 and later appears to be broken. I have
a hackish patch that makes *some* calls work, but it's not an elegant
fix. See https://issues.asterisk.org/view.php?id=15121 f
Hi all,
I've looked at the various variables but can't seem to find a way to
determine which line was picked up in a multi-line ring.
For example, in this excerpt from my asterisk logging:
-- Executing [5558280...@inbound:52] Dial("SIP/proxy3-05ac9180",
"SIP/1555...@proxy1&SIP/1555...@
On Wed, Jun 24, 2009 at 09:20:44PM +0200, jonas kellens wrote:
> I wonder if there is a GUI that does not change the underlying hand-made
> configuration ?!
>
> What I'm looking for actually is a GUI for adding a new SIP-client +
> voicemail, so that a company does not have to call me when they hi
i already did that
- Original Message -
From: Danny Nicholas
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Wednesday, June 24, 2009 1:08 PM
Subject: Re: [asterisk-users] video call doesn work
Make sure the video codecs in the xlite setup are also in si
Thanks.
I didn't change anything in my dialplan. I am aware of reloading configuration
:)
My AGIs are copied from a working asterisk install -- the shebang argument is
how I've always done it. Either way, I have tried it without the -q as well,
and that also didn't succeed.
I just tried your t
On Wed, Jun 24, 2009 at 3:43 PM, Leah Newmark wrote:
> Hi,
>
> I'm running asterisk 1.4.22 on a debian server.
> I have php5 installed and it works correctly command line.
> When trying to run a php script via AGI, I get messages such as:
> GI Tx >> I>
> AGI Rx << #!/usr/bin/php5 -q
> AGI Tx >> 510
Make sure the video codecs in the xlite setup are also in sip.conf
(allow=ulaw,alaw,gsm,h263)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of gmail
Sent: Thursday, June 25, 2009 12:57 PM
To: asterisk-users@lists.digium.com
S
i am trying to make a video call on asterisk 1.6 , my configuration is an
- asterisk 1.6 on Centos on virtual machine VmWare
- Xlite softphone one windows xp (the Host operating system)
- X-lite client on another windows XP (the Guest operating system )
i put the paramtervideosupport=yes
Hi,
I'm running asterisk 1.4.22 on a debian server.
I have php5 installed and it works correctly command line.
When trying to run a php script via AGI, I get messages such as:
GI Tx >> I>
AGI Rx << #!/usr/bin/php5 -q
AGI Tx >> 510 Invalid or unknown command
The scripts are completely executable a
Change writeprotect = no to writeprotect = yes.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Wednesday, June 24, 2009 2:21 PM
To: Asterisk Mailing
Subject: [asterisk-users] GUI for Asterisk
I wonde
On Wed, Jun 24, 2009 at 3:20 PM, jonas kellens wrote:
> I wonder if there is a GUI that does not change the underlying hand-made
> configuration ?!
>
> What I'm looking for actually is a GUI for adding a new SIP-client +
> voicemail, so that a company does not have to call me when they hired a ne
I wonder if there is a GUI that does not change the underlying hand-made
configuration ?!
What I'm looking for actually is a GUI for adding a new SIP-client +
voicemail, so that a company does not have to call me when they hired a
new employee.
I don't want a GUI that over-writes my hand-made SIP
On Wed, 24 Jun 2009, David Backeberg wrote:
> On Thu, Jun 18, 2009 at 2:16 AM, DHAVAL
> INDRODIYA wrote:
>>
>> I am using festival as an application but it default voice is not good
>> to hear anybody have solution about better voice in Festival
>
> If you don't like the output of the free TTS,
On Wed, Jun 24, 2009 at 10:58 AM, Grygoriy Dobrovolskyy wrote:
>
>
> 2009/6/24 Senad Jordanovic
>
>> Jay Fenton wrote:
>> > [ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
>> >
>> > Howler Technologies are proud to announce today the launch of
>> > their fully indemnified and highly optim
On Thu, Jun 18, 2009 at 2:16 AM, DHAVAL
INDRODIYA wrote:
> hello All
>
> I am using festival as an application
>
> but it default voice is not good to hear
>
> anybody have solution about better voice in Festival
I'm of the opinion that festival is:
a) pretty good
b) better than it used to be if y
On Wed, Jun 17, 2009 at 7:10 PM, Marshall
Henderson wrote:
> architecture, etc. On a brand new dual or quad core xeon type
> system(quite likely multiple physical CPUs, each with multiple cores),
> And finally, are there any hard or soft limits to be concerned about
> in regards to the number of si
Jeff LaCoursiere schrieb:
> I have a question in to them about how that floating licensing works,
> though. Does that mean that with every call a license check must be made?
> I don't see how it would work otherwise, and that means my whole business
> - every call - is dependant on their licens
On Wed, 24 Jun 2009, Grygoriy Dobrovolskyy wrote:
> 2009/6/24 Senad Jordanovic
>
>> Jay Fenton wrote:
>>> [ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
>>>
>>> Howler Technologies are proud to announce today the launch of
>>> their fully indemnified and highly optimised G.729A solution
On Wed, 24 Jun 2009 15:11:42 + (UTC), Jeff LaCoursiere wrote:
>
>On Wed, 24 Jun 2009, Grygoriy Dobrovolskyy wrote:
>
>> 2009/6/24 Senad Jordanovic
>>
>>> Jay Fenton wrote:
[ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
Howler Technologies are proud to announce today the
The Asterisk Development Team is pleased to announce the release of
DAHDI Linux 2.2.0 and DAHDI Tools 2.2.0. Both releases are available
for immediate download at http://downloads.asterisk.org/pub/telephony
In addition to various bug fixes, these releases include:
* Support for new Xorcom Astriba
2009/6/24 Senad Jordanovic
> Jay Fenton wrote:
> > [ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
> >
> > Howler Technologies are proud to announce today the launch of
> > their fully indemnified and highly optimised G.729A solution
> > for Asterisk, including a unique floating license mo
Jay Fenton wrote:
> [ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
>
> Howler Technologies are proud to announce today the launch of
> their fully indemnified and highly optimised G.729A solution
> for Asterisk, including a unique floating license model.
Why would someone buy it instead o
Hi,
I want to install an Sangoma A200 together with an BRI card.
I would like to use Asterisk 1.4
Are there any howto or tips?
First compile bristuff and after compile wanpipe?
thanks...
___
-- Bandwidth and Colocation Provided by http://www.api-digital
Jay Fenton wrote:
> [ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
>
> Howler Technologies are proud to announce today the launch of
> their fully indemnified and highly optimised G.729A solution
> for Asterisk, including a unique floating license model.
Please do not post advertisements
> - Original Message -
> From: "Dave Fullerton"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, June 23, 2009 8:39 AM
> Subject: Re: [asterisk-users] Minimizing downtime during updates
>
>
>> Karl Fife wrote:
>>> I was about to ask this question when
In the sip conf for the extension make sure you populate the mailbox
option and put in @
Ish
DHAVAL INDRODIYA wrote:
> hi all,
>
> I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed
>
> when i leave voicemail On Asterisk i need MWI Indication on kamailio
> extension
>
> ther
It is a php problem
you need to put a break in your case 2
case 2:break;
case 3:
$mail->AddAddress(User2);
$mail->AddAddress(User1);
break;
Ish
Danny Nicholas wrote:
> First of all, this is a PHP problem, not an asterisk one. Tha
jonas kellens schrieb:
> I want to use JabberSend in my dialplan, but I saw that my Asterisk does
> not support Jabber.
> Also I have nowhere a module res_jabber.so...
>
> So I thought I'd rebuild my Asterisk. In menuselect I saw that
> res_jabber was dependent of 'iksemel' and 'gnutls'.
>
> In m
Hi all,
Before I start with analog GSM gateways I wanted to check if maybe someone
actually got a working combination of chan_mobile and bluez. If you do
please share specifics like versions, phone, BT chipset, any other relevant
info.
Thanks,
Sasa Bobek
_
[ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ]
Howler Technologies are proud to announce today the launch of
their fully indemnified and highly optimised G.729A solution
for Asterisk, including a unique floating license model.
This is the first in a series of products dubbed 'Howlets'
th
Bart Coninckx wrote:
> Hi,
>
> I'm using a ISDN-30 E1 line from KPN Belgium.
>
> The challenge is to get a correct CallerID on outgoing lines.
>
> When I put this in my dialplan:
>
> exten => _0.,1,Set(TEMPVAR=${CALLERID(num):1})
> exten => _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
> exten => _0.,
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten => _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten => _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten => _0.,3,NoOp(${CALLERID(num)})
exten => _0.,4
hi all,
I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed
when i leave voicemail On Asterisk i need MWI Indication on kamailio
extension
there are some methods i tried but still cant get success
All other feature are working fine also try voip-info.org methods
can anybody
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael Maxwell wrote:
> I had the same issue, only from gtalk to asterisk with some connections and
> not others..
>
> Asterisk to gtalk works fine here.
>
> Have you allowed the RTP ports past your firewall?
small update ... with * 1.6.1.1 :
- g
I want to use JabberSend in my dialplan, but I saw that my Asterisk does
not support Jabber.
Also I have nowhere a module res_jabber.so...
So I thought I'd rebuild my Asterisk. In menuselect I saw that
res_jabber was dependent of 'iksemel' and 'gnutls'.
In my yum repositories I can find a gnutls.
Hello,
I need help to use my sangoma card a108d.
I need that another server give me an E1 with a clock.
The server with the sangoma reseive the E1 clock on port1 and is MASTER E1
on port2.
But, I cant receive the clock (I am connected).
Anyone can help me?
Thank you
Cordialeme
David Backeberg writes:
> As I understand it, you have enabled silence suppression, the silence
> is getting suppressed, and that is worse than when you were not using
> silence suppression. So how about not using silence suppression?
Silence suppression isn't just a "break your telephony" optio
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