Dear Folks,
Im looking for a way to detect if an analog line is connected to card or not
(Im using Sangoma A200). Im using the dialtone detection when dialing but
need a way to detect the disconnection of the line when it actually happens.
Anyone have any hints or tricks for this?
Regards.
--
Thank you Alex, I'll handle this programatically if there is no other way.
Best regards,
Patrick
On Thu, Sep 17, 2009 at 07:51, Alex Balashov abalas...@evaristesys.com wrote:
You can set some kind of counter in the dial plan, call an AGI script,
use func_odbc to make database calls, or
On Wed, 16 Sep 2009, Tilghman Lesher wrote:
On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
What g729 module should I download ?
You should download only the licensed g.729 module from Digium, after paying a
$10 license per concurrent user. All other modules have various
On Wed, 16 Sep 2009, Danny Nicholas wrote:
What do you want your message to say? I'd just use busy-pls-hold and the
caller would eventually get the idea that you weren't going to talk to them.
You could also consider these
Off-duty
Not-auth-pstn
Not-taking-your-call
Number-not-answering
On Thu, Sep 17, 2009 at 09:34:56AM +0330, M Shokuie wrote:
Dear Folks,
Im looking for a way to detect if an analog line is connected to card or not
(Im using Sangoma A200). Im using the dialtone detection when dialing but
need a way to detect the disconnection of the line when it actually
It's a bit off topic here (I would ask this on a QM or TB forum), but
basically you redirect each IVR selection to a context where logging happens
and then redirect to the queue.
Just my two eurocents,
l.
2009/9/16 Maria Cristina Bayno falls_m...@yahoo.com
Hello Team,
IVR selection of
On Thu, 17 Sep 2009, Patrick wrote:
I was thinking also to replace the email sent by the voicemail by a php
script. My questions is simple, how do you manage to get the voicemail
variables from the php script ? Or, maybe, you get from stdin the
content of the email that should be send via
Hello Steve,
Thats what I was expecting :-(
I want to send an email in html format as well as sending an SMS to
the mailbox owner using clickatell's api
Any other ways to do this ?
Best regards,
Patrick
On Thu, Sep 17, 2009 at 09:26, Steve Edwards asterisk@sedwards.com wrote:
On Thu, 17
On Wed, 16 Sep 2009, Steve Edwards wrote:
On Wed, 16 Sep 2009, Danny Nicholas wrote:
I'd try this:
- exten = 4000,1,Dial(SIP/4000,20,ikKtT)
- exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
- exten = s-NOANSWER,2,Voicemail(4000)
- exten = s-BUSY,1,Dial(SIP/4001,20,iKkTt)
- exten =
Alex Samad a écrit :
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
exten = _XXX.,20(Start),Set(GROUP()=PSTN)
exten = _XXX.,n,GotoIf($[${GROUP_COUNT(PSTN)}=0]?lineOpen)
On Thu, Sep 17, 2009 at 08:18:13AM +1000, Alex Samad wrote:
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
Asterisk will not send out more than one call on that line.
You want
Jeff LaCoursiere wrote:
The last patch for RPID is marked for 1.4.23.1 (2/10/09) :
https://issues.asterisk.org/file_download.php?file_id=21601type=bug
I've been running it on 1.4.23.1 since Feb.
Thanks!
I'll give it a shot,
Doug
--
Ben Franklin quote:
Those who would give up
Hello team,
Thanks Lenz, we actually did that.
The ivr data capture at our end is working. We only want to capture one row
per call. Is there any idea regarding this?? tHank you so much.
Regards,
Cristina
--- On Thu, 9/17/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
From: Lenz Emilitri
On 09/17/2009 02:52 PM, Gordon Henderson wrote:
On Wed, 16 Sep 2009, Tilghman Lesher wrote:
On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
What g729 module should I download ?
You should download only the licensed g.729 module from Digium, after paying
On Thu, 17 Sep 2009, Steve Underwood wrote:
Where do you think he might live? Antarctica? :-)
Beirut, Lybia.
Gordon
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register
Actually what you mean by cheaper? 10 cents?
===
Live rates at 2009.09.17 12:49:51 UTC
5.99 GBP
=
9.89128 USD
United Kingdom Pounds United States Dollars
1 GBP = 1.65130 USD 1 USD = 0.605584 GBP
I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version
1.400 and I am simply trying to configure into the Extensions.conf
script an entry that will add to the Auto-Attendant a line that will
allow a Caller to enter a 0 (Zero) will then ring the extension(s)
of the Operator to
On Thu, Sep 17, 2009 at 01:27:19PM +0100, Gordon Henderson wrote:
On Thu, 17 Sep 2009, Steve Underwood wrote:
Where do you think he might live? Antarctica? :-)
Beirut, Lybia.
Beirut in Lybia? I thought Lybia only managed to snach Tripoly from
Lebanon.
--
Tzafrir Cohen
This is simple; in the [ivr] context, add
- exten = 0,1,Dial(SIP/238,20,KkTt)
A better way to handle this is to write a key in your asterisk database
and set up a context to allow changing the value
[ivr]
- exten = 0,1,Set(OPEREXT=${DB(Oper/ext)})
- exten = 0,2,Dial(SIP/${OPEREXT},20,KkTt)
Else
On Thursday 17 September 2009 01:52:26 Gordon Henderson wrote:
On Wed, 16 Sep 2009, Tilghman Lesher wrote:
On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
What g729 module should I download ?
You should download only the licensed g.729 module from Digium, after
paying a
When I call inbound with a cell phone (via SIP PSTN trunk) some of my
prompts the first word is cut off. I'm assuming the prompt is needing
to be transcoded on the fly and it's not getting transcoded fast
enough. I did a file convert to create gsm versions (currently they
are referenced
On Thursday 17 September 2009 02:52:42 Patrick wrote:
I want to send an email in html format as well as sending an SMS to
the mailbox owner using clickatell's api
Any other ways to do this ?
Use the externnotify option in voicemail.conf. The arguments to this
program are the context,
On Thu, 17 Sep 2009, Hristo Benev wrote:
Actually what you mean by cheaper? 10 cents?
As I live in the UK, it's more. By the time I add on my greedy banks
conversion fees, etc. then if I'm honest and add on any import
duties/customs, etc. it's way more than just 10 cents.
And I know you'd
On Thu, 17 Sep 2009, Tzafrir Cohen wrote:
On Thu, Sep 17, 2009 at 01:27:19PM +0100, Gordon Henderson wrote:
On Thu, 17 Sep 2009, Steve Underwood wrote:
Where do you think he might live? Antarctica? :-)
Beirut, Lybia.
Beirut in Lybia? I thought Lybia only managed to snach Tripoly from
James Hankins wrote:
When I call inbound with a cell phone (via SIP PSTN trunk) some of my
prompts the first word is cut off. I'm assuming the prompt is needing
You need to add a Wait(1) after the Answer()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
Just tried, perfect! Thanks.
Jim
On Sep 17, 2009, at 9:56 AM, Doug Lytle wrote:
James Hankins wrote:
When I call inbound with a cell phone (via SIP PSTN trunk) some of my
prompts the first word is cut off. I'm assuming the prompt is
needing
You need to add a Wait(1) after the
James Hankins wrote:
When I call inbound with a cell phone (via SIP PSTN trunk) some of my
prompts the first word is cut off. I'm assuming the prompt is needing
On Thu, 17 Sep 2009, Doug Lytle wrote:
You need to add a Wait(1) after the Answer()
Or answer(1000).
--
Thanks in advance,
Hi all,
Several years ago there was a thread on this list about the behavior
of Asterisk when there was an empty display-name field in the SIP FROM
header:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg142835.html
It seems as if the thread was not answered, so allow me to
On Thursday 17 September 2009 02:52:42 Patrick wrote:
I want to send an email in html format as well as sending an SMS to the
mailbox owner using clickatell's api
Any other ways to do this ?
On Thu, 17 Sep 2009, Tilghman Lesher wrote:
Use the externnotify option in voicemail.conf. The
Hellos
I am using freepbx and asterisk.
I am writing an AGI script to edit the values in findmefollow table. The
script will enable users to delete and add follow me numbers from their
phones. I want it to enable users enable/disable follow me.
I can't seem to find a value in the database that
On Thursday 17 September 2009 09:13:20 Steve Edwards wrote:
On Thursday 17 September 2009 02:52:42 Patrick wrote:
I want to send an email in html format as well as sending an SMS to the
mailbox owner using clickatell's api
Any other ways to do this ?
On Thu, 17 Sep 2009, Tilghman
Well, why not disable it from the GUI and see what changes -- this is
sort of the wrong list, but maybe someone knows more fully.
James Mutuku listmut...@gmail.com wrote:
Hellos
I am using freepbx and asterisk.
I am writing an AGI script to edit the values in findmefollow table. The
On Thu, 17 Sep 2009, Tilghman Lesher wrote:
On Thursday 17 September 2009 01:52:26 Gordon Henderson wrote:
On Wed, 16 Sep 2009, Tilghman Lesher wrote:
On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
What g729 module should I download ?
You should download only the licensed
Steve Edwards wrote:
Or answer(1000).
I'd prefer this route! Thanks.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and
On Thu, Sep 17, 2009 at 10:50 AM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
On Thu, 17 Sep 2009, Tilghman Lesher wrote:
The free one or the Howlets one?
However I can't see how the binary blobs of patented code which digium
sells doesn't voilate the GPL
Steve Edwards escribió:
James Hankins wrote:
When I call inbound with a cell phone (via SIP PSTN trunk) some of my
prompts the first word is cut off. I'm assuming the prompt is needing
On Thu, 17 Sep 2009, Doug Lytle wrote:
You need to add a Wait(1) after the Answer()
Yes, Steve could write a book. I would probably buy The Luddites guide to
Asterisk.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
Sent: Thursday, September 17, 2009 10:14 AM
To: Asterisk Users Mailing
I'm having a little problem with voicemail. Actually I'm not getting the
ability to leave a voicemail-message.
This is part of the dialplan :
exten = s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1})
exten = s,n,NoOp(${ar...@boxes)
exten = s,n,Voicemail(${ar...@boxes)
On Thu, 2009-09-17 at 17:31 +0200, jonas kellens wrote:
vm-intro is an empty file. I deleted the original and replaced it with
a touch vm-intro.gsm.
I'm curious as to why you did this. Why didn't you simply pass the 's'
option to the VoiceMail() application to have it skip the introductory
At 06:49 AM 9/17/2009, you wrote:
When I call inbound with a cell phone (via SIP PSTN trunk) some of my
prompts the first word is cut off. I'm assuming the prompt is needing
to be transcoded on the fly and it's not getting transcoded fast
enough. I did a file convert to create gsm versions
On Thursday 17 September 2009 10:01:46 Moises Silva wrote:
On Thu, Sep 17, 2009 at 10:50 AM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
On Thu, 17 Sep 2009, Tilghman Lesher wrote:
The free one or the Howlets one?
However I can't see how the binary
I have tried all that. I just can't trace the value. this maybe the wrong
list. I just thought someone might know
On Thu, Sep 17, 2009 at 5:39 PM, cov...@ccs.covici.com wrote:
Well, why not disable it from the GUI and see what changes -- this is
sort of the wrong list, but maybe someone knows
Hi,
This week we are pleased to welcome Andy Abramson
(http://andyabramson.blogs.com/voipwatch/) as our guest. Andy is one
of the most avid observers of the world of VoIP, from Asterisk and its
variations to all kinds of ramifications of VoIP. I'm sure we'll pass
a lot of the VoIP News in review
Close enough. Digium doesn't own the Asterisk code, but it possesses
enough
of a copyright interest in the code, as well as licenses from all
contributors, in order to be able to make that exception.
Ah, yeah, I stand corrected. I should have not used own. For the casual
reader, the
Nobody on this ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: September-16-09 7:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] H323 RTP Transmission error of packet
Using H323 to reach
Jared Smith jsm...@digium.com writes:
Again, the emphasis on the dCAP exam is real-world knowledge of how to
build a simple small-business PBX with Asterisk. If you've used
Asterisk in a professional capacity, it should be very straightforward
to pass the practical portion of the exam.
I
Benny Amorsen wrote:
Jared Smith jsm...@digium.com writes:
Not that I would ever consider taking an exam like that, but I have been
using/configuring asterisk since nearly the beginning of this mailing
list, and I have never touched dahdi or polycom. Someone should still be
able to pass
Since Digium's contribution to Asterisk (hardware-wise) is Analog DAHDI
cards, this makes sense (to me). I suppose they could make a DCAP exam that
just used SIP trunks and softphones, but then that would just be GCAP
(Generic Certified Asterisk Professional) or SCAP (SIP Certified Asterisk
Hi people, I have the following dialplan:
[context]
exten = s,1,Noop(Start)
...
exten = h,1,Noop(Ending)
exten = h,n,DEADAGI(finconf.php,${ARG1},${ARG2})
When it is running, the asterisk gives the following error:
-- Launched AGI Script /var/lib/asterisk/agi-bin/finconf.php
==
Either the file is missing something like #!/usr/bin/php or there is an
error in what the file is trying to access
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Thursday, September 17, 2009 2:58 PM
To:
1) does the file exist
2) is it chmod'd to 755 (not sure if this matters though)
3) do you have something like #!/usr/bin/php at the start of the php file?
Cheers
Geraint
2009/9/17 Anahi Ludueña a_ludu...@hotmail.com
Hi people, I have the following dialplan:
[context]
exten =
On Thursday 17 September 2009 15:06:28 Geraint Lee wrote:
1) does the file exist
2) is it chmod'd to 755 (not sure if this matters though)
3) do you have something like #!/usr/bin/php at the start of the php file?
4) Is the file in MS-DOS format (i.e. do you have \r\n at the end of every
line,
Aloha,
I too am running into a similar problem. I have version 1.6.1.5 that
works fine, however 1.6.1.6 does not.
I have both requirecalltoken=no calltokenignore=xxx.xxx.xxx.xxx in
iax.conf, but I am still getting this error message.
[Sep 17 09:54:23] ERROR[32335]: chan_iax2.c:4529
Thanks for the answers!
The file didn't have the first line!
#!/usr/bin/phpBye!
Anahi Ludueña
From: tles...@digium.com
To: asterisk-users@lists.digium.com
Date: Thu, 17 Sep 2009 15:59:21 -0500
Subject: Re: [asterisk-users] DeadAgi
On Thursday 17 September 2009 15:06:28 Geraint
On Thu, 17 Sep 2009, Anahi Ludue?a wrote:
Thanks for the answers!
The file didn't have the first line!
#!/usr/bin/php
Glad you found the answer. However...
The command ls -l returns:
-rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php
Having an executable with 777 permissions
Hello -
I am fairly new to Asterisk, but we have a fully operational system with
very few hiccups. Much of that is because of this list. Thanks.
My question is this -
We have assigned MeetMe conference IDs to all of our employees. We then
setup a TN to accommodate the MeetMe() app.
I am working on updating to 1.6.1.6 and if I have res_snmp.so
auto-loading on CentOS 5.3 Asterisk Seg faults every time. I can load
the module manually after the initial startup. I am starting to dig
into it further and will open a ticket, just wanted to see if anyone
else knew of any issues off
On 18/09/09 7:12 AM, jon pounder wrote:
Benny Amorsen wrote:
Jared Smithjsm...@digium.com writes:
Not that I would ever consider taking an exam like that, but I have been
using/configuring asterisk since nearly the beginning of this mailing
list, and I have never touched dahdi or polycom.
Hi guys,
Anyone done this with CentOS and asterisk 1.4?
thanks
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users
At 7:16 AM on 17 Sep 2009, Patrick wrote:
I've one SIP trunk that support multiple DID. Only the trunk is
documented in sip.conf (called DID is taken from the sip-header in
real time).
I would like to limit the number of simultaneous calls on each DID. Is
there a way to achieve this ?
I
Aloha,
I'm finishing up the final touches on this install, and have run into an
odd problem.
I can't seem to get Caller ID on the analog phone lines working. It's a
Digium AEX 410 card.
I have Verbose set and a line to print the CID:
I have usecallerid=yes and callerid=asreceived set in both
I have with CentOS 5.3 and custom 1.6.1.6 RPMs. If you use RPMs for
the installation of Asterisk then it's really easy. As for the
Kickstart, if you haven't used it much here I did a quick write-up
with example script here: http://thurmantech.com/node/3
Either use RPMs and add them to the
Hi, All,
I always get 404 respond when I send SUBSCRIBE to asterisk. Does anybody
know why?
Message flow is as follows:
SUBSCRIBE sip:1...@192.168.1.32:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.36:5060;rport;branch=z9hG4bKPjdf42536ef66447529e3da381f3556c1b
Max-Forwards: 70
From: 1003
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