jordan pan schrieb:
Hi everyone,
when i use the automated dial out,I found that once the zap
answerd,the contex will be exectued, but i don't hope do it ,i hope
when extern phone answered ,then ,the context will be exectued.
Anyone can help me solve the problem!
the call file is:
2008/12/23 Yehavi Bourvine yehavi.bourv...@gmail.com
I have one ST2030 bought for testing. Indeed it has a very intuitive user's
interface, bue I've found two drawbacks:
- Its sound quality has some place to be improved...
- It has no RPID support (displaying the name of the called
I'll have to run some TCPDUMP to see what happens. I'll also try this with
OpenSIPS where there is more flexbility with the header fields.
Thanks, __Yehavi:
2008/12/23 Olivier oza-4...@myamail.com
2008/12/23 Yehavi Bourvine yehavi.bourv...@gmail.com
I have one
On Tue, Dec 23, 2008 at 1:48 AM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
I have one ST2030 bought for testing. Indeed it has a very intuitive user's
interface, bue I've found two drawbacks:
Its sound quality has some place to be improved...
It has no RPID support (displaying the name
Hi!
I wonder why users.conf generates a SIP user and a SIP peer? Why is it
not possible to set type=... in users.conf? (Asterisk 1.4.22)
thanks
klaus
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
It's all ball bearings these days
On Tue, Dec 23, 2008 at 4:35 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Hi!
I wonder why users.conf generates a SIP user and a SIP peer? Why is it
not possible to set type=... in users.conf? (Asterisk 1.4.22)
thanks
klaus
On Mon, Dec 22, 2008 at 09:33:18AM -, Andrew Thomas wrote:
You don't really need to use any local MTA if you use the sendEmail
script.
I got it from - http://www.caspian.dotconf.net/menu/Software/SendEmail/
Which is essentially the same as using ssmtp / esmtp / nullmailer
(non-queuing
On Mon, Dec 22, 2008 at 10:46:46AM +0100, Olivier wrote:
Hi Andrew,
2008/12/22 Andrew Thomas a...@datavox.co.uk
JFYI - I run (successfully) agx-addons with 1.4.22 and Etch.
Make sure you have the right version of SpanDSP installed (as well as the
tiff libraries).
which are
On Mon, Dec 22, 2008 at 09:04:13AM -0600, Wesley Haut wrote:
Hi all,
I know I'm probably stirring up a hornet's nest with this question/comment
but I've spent the last few days working on a PHP-based class for the
manager interface
Isn't there one already?
as we're preparing for a pretty
On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote:
Hey everyone,
A while back I worked on a project to measure call quality. I've
finally gotten around to releasing it and I'm calling it recqual (Real
Call Quality). There isn't much to it and it should be considered
On Tue, Dec 23, 2008 at 10:35:19AM +0100, Klaus Darilion wrote:
Hi!
I wonder why users.conf generates a SIP user and a SIP peer? Why is it
not possible to set type=... in users.conf? (Asterisk 1.4.22)
users.conf is a hack to generate a typical Asterisk configuration
easily.
So I figure
On Mon, Dec 22, 2008 at 03:30:02PM -0500, Noah Miller wrote:
Hi All -
I'm wondering if anybody has IMAP Voicemail AND the directory working
together. I haven't had any success. IMAP voicemail works fine, but
when it's active, the Directory does not work. The problem seems to
be with
Thanks all for your replies.
I have an aastra 9133i here for testing and am getting a polycom 320 to try
out.
But today, I got my hands on an older Cisco 7912G with SIP software
installed. It connected fine to the Asterisk box, works with the PoE stuff
I have, sounds good and doesn't seem to
Hi all,
anyone have any experience regarding query whether our sip accounts
registered (online) or not registered (offline) from out of asterisk with
mysql or another tool. My goal is taking this information with query and
put it to my intranet to check my users.
any help would be appreciated
On Tue, Dec 23, 2008 at 09:06:26AM +0100, Stefan Schmidt wrote:
jordan pan schrieb:
Hi everyone,
when i use the automated dial out,I found that once the zap
answerd,the contex will be exectued, but i don't hope do it ,i hope
when extern phone answered ,then ,the context will
We have one 7912 which we bought for evaluation. The main drawback is that
it has hands free speaker but no microphone.
__Yehavi:
2008/12/23 Mikel Lindsaar raasd...@gmail.com
Thanks all for your replies.
I have an aastra 9133i here for testing and am getting a polycom
On Tue, Dec 23, 2008 at 1:28 AM, Danny Nicholas da...@debsinc.com wrote:
What does your extensions.conf look like for this call? If you can
insert a ww into your Dial command (ie, change 18005551212 to ww18005551212)
this may improve your dialing behavior.
In an attempt to isolate the
On Tue, Dec 23, 2008 at 5:23 PM, yavuzhan canli yca...@tekfen.com.trwrote:
Hi all,
anyone have any experience regarding query whether our sip accounts
registered (online) or not registered (offline) from out of asterisk with
mysql or another tool. My goal is taking this information with
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mikel Lindsaar wrote:
What does putting ww at the front do?
Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial.
(It may be a 1/4 second each 'w')
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
On Tue, Dec 23, 2008 at 11:01 PM, Yehavi Bourvine yehavi.bourv...@gmail.com
wrote:
We have one 7912 which we bought for evaluation. The main drawback is that
it has hands free speaker but no microphone.
That's true. But we will be getting higher models for the speaker function.
Did you find
On Wed, Dec 24, 2008 at 12:02 AM, Barry L. Kline blkl...@attglobal.netwrote:
What does putting ww at the front do?
Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial.
(It may be a 1/4 second each 'w')
I thought so, in that case, it is not the problem here.
My problem
Isn't there one already?
Yeah, but none of them have worked for me...maybe their way of doing things
is just different from my approach but I wasn't happy with any of the
existing classes. I wasn't planning on releasing my code to the wild (I'm
not a programmer by trade I just play one on TV).
I have a TE210P digium card that has 2 E1/T1 ports.
the code in my extensions.conf file for span 1 is :
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1; Trunk interface
TRUNKX=Zap/g2
On Dec 22, 2008, at 10:38 PM, Martin Lima wrote:
On Thursday 18 December 2008, Justin Phelps wrote:
I've been struggling with an ongoing problem the last month.
Here is the layout of the wiring:
T1 from ISP DiTech Echo Cancel device Voice Channel-Bank
(same) T1 from ISP (same) DiTech
Hi Tzafrir -
I'm wondering if anybody has IMAP Voicemail AND the directory working
together. I haven't had any success. IMAP voicemail works fine, but
when it's active, the Directory does not work. The problem seems to
be with libc-client. Specifically, asterisk is not able to access the
Dear Sir,
I used several other Softphones like Skype and they are facing the same
problem...It seems that the issue is global du to an undersea cable cut
Regards
On Mon, Dec 22, 2008 at 9:07 PM, michel freiha mich...@gmail.com wrote:
Hi all,
Sometimes when making a PC to PSTN call through
On Tue, Dec 23, 2008 at 10:13:12AM -0500, Noah Miller wrote:
Hi Tzafrir -
I'm wondering if anybody has IMAP Voicemail AND the directory working
together. I haven't had any success. IMAP voicemail works fine, but
when it's active, the Directory does not work. The problem seems to
be
On Tue, Dec 23, 2008 at 4:40 AM, Steve Totaro
stot...@first-notification.com wrote:
It's all ball bearings these days
What is the deal with Fletch quotes these days? Don't get me wrong, I
appreciate them but I'm starting to wonder where this is all coming
from.
I *think* it's because Fletch
On Tue, Dec 23, 2008 at 6:31 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote:
Hey everyone,
A while back I worked on a project to measure call quality. I've
finally gotten around to releasing it and I'm calling it
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is
This has been on my ToDo list far too long.
I have a small call-center setup, with basic
time of day/day of week validation before putting
callers in the queues.
With the holidays upon us, I need to add check to
see if 'today' is a holiday so I do not put callers
in unmanned queues. Due to how
I have two offices sharing a phone system. They also share a common
internal context because all of the employees of the second office also
work for the first office. Each office has 4 outside lines and I have
defined 2 channel groups in my zapata.conf. The second office needs all
of their
On Tuesday 23 December 2008 11:47:08 Brent Davidson wrote:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the
Brent Davidson schrieb:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the
On Tuesday 23 December 2008 12:11:41 Dan Austin wrote:
This has been on my ToDo list far too long.
I have a small call-center setup, with basic
time of day/day of week validation before putting
callers in the queues.
With the holidays upon us, I need to add check to
see if 'today' is a
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file
Philipp Kempgen wrote:
Brent Davidson schrieb:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning:
On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote:
Hey everyone,
A while back I worked on a project to measure call quality. I've
finally gotten around to releasing it and I'm calling it recqual (Real
Call Quality). There isn't much to it and it should be considered
Brent Davidson wrote:
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]:
On Tue, Dec 23, 2008 at 2:59 PM, Mindaugas Kezys mke...@gmail.com wrote:
Looks very interesting. After reading all available info I have two
questions before testing:
1. Who/what answers the calls at the other end? I guess real live traffic
should be sent through this Asterisk server?
2.
I have been trying to figure out how the * works when in the Directory
(dial-by-name). When I press * (which is supposed to exit the directory) I
end up somewhere which I never specified. It seems like Asterisk just
picked that place to go, because I never specified it.
The wiki is no help
On Tuesday 23 December 2008 14:49:52 Mike wrote:
I have been trying to figure out how the * works when in the Directory
(dial-by-name). When I press * (which is supposed to exit the directory) I
end up somewhere which I never specified. It seems like Asterisk just
picked that place to go,
On Mon, Dec 22, 2008 at 5:37 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hey everyone,
A while back I worked on a project to measure call quality. I've
finally gotten around to releasing it and I'm calling it recqual (Real
Call Quality). There isn't much to it and it
Mike wrote:
I have been trying to figure out how the * works when in the Directory
(dial-by-name). When I press * (which is supposed to exit the
directory) I end up somewhere which I never specified. It seems like
Asterisk just picked that place to go, because I never specified it.
Dave Fullerton wrote:
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?
-Dave
I'm running 1.4.21.2 and I can't upgrade until
On Tue, 23 Dec 2008, Brent Davidson wrote:
Dave Fullerton wrote:
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?
-Dave
I'm
Not the most elegant but since I have a generic context for my IVRs I
simple check the date there.
exten = s,n,GotoIfTime(*|*|1|jan?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|10|apr?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|25|may?closed-holiday|1)
exten =
On Tue, Dec 23, 2008 at 4:02 PM, Atis Lezdins a...@iq-labs.net wrote:
Hi,
This is good idea, and i will probably try it out someday next year
(too busy completing my business requirements :)
Luckily next year is just over a week away. We won't have to wait
that long ;).
I took a look at
On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote:
Unfortunately 1.4.22 no
longer has Zaptel. :(
Asterisk 1.4.22 builds with both Zaptel and DAHDI.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
On Tue, Dec 23, 2008 at 11:17 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
This is true, however, I wasn't very excited about any other debug
messages that might get printed with debug 1. I knew I only needed
the endpoint RTP address, so I just removed the if. Of course you
Tzafrir Cohen wrote:
On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote:
Unfortunately 1.4.22 no
longer has Zaptel. :(
Asterisk 1.4.22 builds with both Zaptel and DAHDI.
I spent several hours trying to make it work yesterday and it just
wouldn't. I kept getting an
Thanks, to you and Mark, for the quick reply. I used to rely on the Wiki
but it seems I shouldn't
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Tuesday, December 23, 2008
Jeff LaCoursiere wrote:
On Tue, 23 Dec 2008, Brent Davidson wrote:
Dave Fullerton wrote:
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk
We chose to use a mySQL database to store the holiday information.
When a call is answered we query the database to see if there is a
holiday greeting recorded, if so we play the indicated greeting,
otherwise play the default menu greeting. (We do our dialplans in AEL)
context
On Tue, Dec 23, 2008 at 04:01:24PM -0600, Brent Davidson wrote:
Tzafrir Cohen wrote:
On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote:
Unfortunately 1.4.22 no
longer has Zaptel. :(
Asterisk 1.4.22 builds with both Zaptel and DAHDI.
I spent several hours
Tilghman wrote:
Astdb is a nice idea. Something along the lines of:
GotoIf(0${DB(holiday/${STRFTIME(,,%Y-%m-%d)})}?holiday,s,1)
would work. Holidays are evaluated as 01, which is true.
Anything not in the database would be evaluated as 0, which
is false. This will work both for holidays
Tzafrir Cohen wrote:
What error message from where?
With Zaptel the echo canceller settings are global (that is: one
hard-coded echo canceller). With DAHDI there are echo canceller modules
and you can (and actually need to) set them per-channel.
It might have something to do with the
[Dec 23 17:58:49] ERROR[3091]: chan_dahdi.c:8413 dahdi_pri_error: XXX
Missing handling for mandatory IE 12 (cs0, Connected Number) XXX
I am seeing the above error on DAHDI 2.1.0, asterisk 1.4.22 and libpri 1.4.7
I am using a TE120P card.
I am also getting this VERY frequently:
-- Channel
On Dec 23, 2008, at 4:49 PM, Mike wrote:
Thanks, to you and Mark, for the quick reply. I used to rely on the
Wiki
but it seems I shouldn't
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of
2008/12/3 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Tuesday 02 December 2008 12:22:16 Dave Fullerton wrote:
Is anyone else having difficulty compiling 1.6.0.2?
I'll get a new release candidate out either this afternoon or tomorrow;
I'm currently working on ensuring that 1.6.0.3
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