Set your PRI for intensive debugging for a few outbound calls and trim it to
show the relevant info.
Intense PRI debugging will probably give you the answer right off the bat.
Check your zapata.conf settings:
*restrictcid*: (PRI channels only) This option has effect only when
hidecallerid=no. If
Steve .T sir, All
It is working sir . Thank you very much sir .
Thankyou for all support to me .
sorry for the my language bcz i have my communication problem.
On Wed, May 11, 2011 at 11:30 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
Set your PRI for intensive debugging for a few
No problem, what was the solution?
On Wed, May 11, 2011 at 2:14 AM, mahesh katta maheshka...@flexydial.comwrote:
Steve .T sir, All
It is working sir . Thank you very much sir .
Thankyou for all support to me .
sorry for the my language bcz i have my communication problem.
On Wed, May 11,
Sir,
I set the below configured in Zapata.conf file. and A .J given Dialplan .
that's it is working now
hidecallerid=no
restrictcid=yes
On Wed, May 11, 2011 at 11:55 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
No problem, what was the solution?
On Wed, May 11, 2011 at 2:14 AM,
Thanks Matt,
can I transform the format in dial plan?
On Wed, May 11, 2011 at 11:24 AM, Matt Riddell li...@venturevoip.comwrote:
On 11/05/11 3:11 PM, John Wu wrote:
Hi Enrico
thanks I do what u said but meet this problem:
[May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File
Hi,
my obd calls to all Idea mobile numbers drops after few sec.
where as, with same configuration , and making obd on to pri it works
properly.
Also i had words with Idea team, they says release is from my point code,
whereas on debugging at my asterisk server i received release from their
Wow! How self-promoting was that post?
As for a simple 'that worked' post - as others have already pointed out
before you, it's not for self-gratification - it's to help anyone else
who has the same/similar problem. I used the list archives quite a lot
in my early days - and having the last post
On Wed, May 11, 2011 at 6:47 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
I don't need a public or private Thank You When I was posting all the
time, I figured the ratio of Thank you emails to silence to be about 20 to
1, maybe as high as 50 to 1.
I agree with the others who are
Thanks Matt
the problem is solved.
On Wed, May 11, 2011 at 11:24 AM, Matt Riddell li...@venturevoip.comwrote:
On 11/05/11 3:11 PM, John Wu wrote:
Hi Enrico
thanks I do what u said but meet this problem:
[May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File
fail.wav does not
On Wednesday 11 May 2011, mahesh katta wrote:
Sir,
I set the below configured in Zapata.conf file. and A .J given Dialplan .
that's it is working now
hidecallerid=no
restrictcid=yes
Glad you got it all sorted -- I was going to suggest a few more things you
could try this morning, but got
On Wed, May 11, 2011 at 3:55 AM, randulo rand...@randulo.com wrote:
On Wed, May 11, 2011 at 6:47 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
I don't need a public or private Thank You When I was posting all the
time, I figured the ratio of Thank you emails to silence to be about
On Wed, May 11, 2011 at 12:48 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Thanks Randulo,
I am surprised you noticed that.
I truly give thanks to all productive members of the Asterisk community.
Second that!
Would you say that I am a productive member of the list and go pretty
On Tue, May 10, 2011 at 8:30 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
+1 from me too. The other thing is that when you answer
to say the problem has been solved this goes into the archives meaning
that people can use Google to answer their own questions
Hello List,
Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the
same.
Dialplan looks like
Exten = 100,1,MeetMe(100,dmF)
Sip.conf
dtmfmode=info
Regards,
Rajib
Hi Rajib,
There is nothing like that Asterisk is blocking an audio if you use without
F it gives you and audio or not.
cheers
Dhaval
On Wed, May 11, 2011 at 5:34 PM, Deka, Rajib IN MAA SL
rajib.d...@siemens.com wrote:
Hello List,
Asterisk is blocking audio if ‘F’ flag is enabled in
Let's not get in to to pissing contest. I am not new to this list (jfyi
- I am also a dCAp). I do know who you are (and couldn't care less
anymore). I, also, have paying customers (but don't feel the need to
gloat about it in here). I am not pretending to know you - as I don't
know you on a
It's working now. I removed the 'm' flag from the meetme Dialplan. It was my
mistake. Asterisk is working fine.
Exten = 100,1,MeetMe(100,dF)
Regards,
Rajib
From: Deka, Rajib IN MAA SL
Sent: Wednesday, May 11, 2011 5:35 PM
To: 'asterisk-users@lists.digium.com'
Hi Dhaval,
Thanks for your much appreciable reply.
Sorry for late reply as I was out of office.
We considered the situation that pending queue call cannot be retrieved during
failover, and hence it's ok with us if we loose the calls also.
Regards,
Rajib
Date: Wed, 4 May 2011 14:15:59 +0530
Jealous much?
Your logic fails you. If you didn't want to get into a pissing match then
why continue after your first statement.
dcap means nothing to me, it is like having your A+ cert but good for you.
You were the one that brought up paying customers. I simply stated that by
helping people
On 05/11/2011 09:29 AM, Steve Totaro wrote:
You must have a reading compression problem.
I would love to bzip2 or gzip the reading process.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web:
Alex, thanks for the laugh.
I have a wireless keyboard and the batteries are dying. I have been lazy
and not picked up some AAAs.
I have been using spell check to help. At least the wrong word was spelled
correctly, lol.
Or he is not really reading what I wrote which was along the same lines
Snore...
Now move along please...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: 11 May 2011 14:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I think he does have a very lossy reading compression problem though. It
actually works both ways.
On Wed, May 11, 2011 at 9:32 AM, Alex Balashov abalas...@evaristesys.comwrote:
On 05/11/2011 09:29 AM, Steve Totaro wrote:
You must have a reading compression problem.
I would love to bzip2
Exactly what I would expect from someone that doesn't have a leg to stand
on.
You move along, I am the one with the guns.
Thanks,
Steve Totaro
On Wed, May 11, 2011 at 9:41 AM, Andrew Thomas a...@datavox.co.uk wrote:
Snore...
Now move along please...
--
Hi again,
no one got an idea? :-( Or did my request not make any sense? Or is the
answer to obvious that no one bothers to reply? :-)
Thanks again!
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command
On 5/10/2011 10:38 AM, Asterisk Development Team wrote:
Below is a sample of the issues resolved in this release:
[...]
For a full list of changes in this release candidate, please see the ChangeLog:
I'm a bit confused about this release (and previous releases on the 1.8
track) so please
On 11-05-11 10:29 AM, Jeremy Kister wrote:
I'm a bit confused about this release (and previous releases on the 1.8
track) so please bare with me.
I viewed the ChangeLog, but I don't see any of the 'sample issues'
listed. why is that ? I would expect to see the 'sample issues' listed
after
On 11-05-11 10:46 AM, Paul Belanger wrote:
On 11-05-11 10:29 AM, Jeremy Kister wrote:
I'm a bit confused about this release (and previous releases on the 1.8
track) so please bare with me.
I viewed the ChangeLog, but I don't see any of the 'sample issues'
listed. why is that ? I would expect
snip
[Danny Nicholas]
Paul, this is probably a dumb question, but why are some (or is it all and
I just don't notice it) modules fundamentally changed from release to
release (or version to version)? As a C-dabbler, it seems to me that if I
do gcc app_voicemail.c (using voicemail as an
Hi,
I'm trying to add modules compiled from source into a rpm install of asterisk
(from digium) on centos and asterisk complains that its not compiled with same
options so it won't load it
I know I could install the entire thing from source but for other reasons I
would like to keep the main
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of isr...@gmail.com
Sent: Wednesday, May 11, 2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] With what
danny,
not that it matters, but I agree. if the design is a good design, it
would not have to be redesigned on every release. in fact, the modules
template should also follow this philosophy that way you can concentrate
on adding functions and not the design...
sometimes, it is smarter to
Wow...somehow this turned into a something so much darker than the original
intent*sits back and watches the show*
Thanks guys, that little mini bonfire made an otherwise boring day into an
entertaining Asterisk-Users version of WWE Raw.
Cheers!
Sherwood McGowan
--
Try reading up on Local channels, it will accomplish everything you wish.
On Wed, May 11, 2011 at 8:59 AM, Markus unive...@truemetal.org wrote:
Hi again,
no one got an idea? :-( Or did my request not make any sense? Or is the
answer to obvious that no one bothers to reply? :-)
Thanks
Yeah, I am not sure why dude went on the offensive. Got emotional but could
not produce a single example of the name calling and insults he was hurling
at me.
Here is an email I received a very short time ago. Sender and company's
name have been removed.
*| to Steve *
*show details 9:15 AM (2
On 05/11/2011 08:39 AM, asterisk-users-requ...@lists.digium.com wrote:
Snore...
Now move along please...
OK, but how about you respecting some basic mailing list etiquette and
not quoting the entire thread in your posts so that those of us who wade
through these messages as digests don't
Hi List
This may be a silly question by web searches etc don't seem to answer it.
Is there a CLI command to display ALL channel variables - standard and user
created - for a specific channel?
something like show channel SIP/Test123 all
I'm using Version 1.4.33.1
PG
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Paddy Grice
Sent: Wednesday, May 11, 2011 11:49 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CLI - displaying all channel variables
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, May 11, 2011 10:55 AM
To: pa...@wizaner.com; Asterisk Users Mailing List -Non-Commercial
Discussion
Subject: Re:
Seems I have upset the God that is Steve Totaro!
You want an example? OK - your last post. Has nothing to do with the
thread (or our 'discussion') but yet you chose to post it as yet another
self pat-on-the-back! I could produce a lot more - but you now bore me.
You know it must be so hard
Thanks Danny - That displays user created variables - by user this could
be application like dial but not the predefined channel variables.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent:
You should probably use an AGI to get this rather than depending on CLI
commands. It's possible that you could do a bash AGI and call that from CLI
but that's not something I've dabbled with.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
On Wed, 11 May 2011, Eric Wieling wrote:
Generally you should insert a Noop in the dialplan to examine variables.
Noop(EXTEN is ${EXTEN}) for example.
The 'verbose()' application would be an example of 'better practices.'
It's function is obvious rather than just a convenient side-effect.
I am not upset in least, well I am but that's because I own thousands of
ounces of silver bullion and I am watching in get pummeled again. Good
thing I bought the bulk of it when it was only $12 an ounce.
http://www.kitco.com/charts/livesilver.html
You are an angry person and it is sad.
It is
Hi all,
I would like to track/store concurrent call usage per user by
day/week/month and get server totals by day/week/month. Google comes up with
mostly info regarding concurrent call limits, though my goal is to calculate
actual concurrent channel usage and add it into reporting. I'm using
On 11-05-11 12:57 PM, Skyler wrote:
I would like to track/store concurrent call usage per user by
day/week/month and get server totals by day/week/month. Google comes up with
mostly info regarding concurrent call limits, though my goal is to calculate
actual concurrent channel usage and add
Hi,
I did replace an old asterisk box by a new shiny one (2 BRI ports used
on a quad port card - BeroNet PCI Express)
I noticed a message in the logs that puzzles me:
May 11 19:10:02 kernel: dahdi: Master changed to B4/0/1
May 11 19:10:08 kernel: wcb4xxp :05:04.0: PCI INT A disabled
May
You can use the manager api (interface) and poll that info and then
store it in a MYSQL table etc.
You can do this outside asterisk,even from a different machine using
your preferred dev language as there are manager libraries/bindings for
most major dev languages
'Actual' is the key word
Hello,
I'm interested in knowing if anyone out there has successfully connected
Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that
we put in an Asterisk install, one of their sister companies who we don't
control is putting in a Cisco UC 560. From my looking I think it can
On 05/11/2011 01:30 PM, Darrin Henshaw wrote:
I'm interested in knowing if anyone out there has successfully
connected Asterisk to a Cisco UC 560 via SIP trunking? We have a
client of ours that we put in an Asterisk install, one of their
sister companies who we don't control is putting in a
On Wed, May 11, 2011 at 12:30 PM, Darrin Henshaw
darrin.aster...@gmail.comwrote:
Hello,
I'm interested in knowing if anyone out there has successfully connected
Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that
we put in an Asterisk install, one of their sister
On 11-05-11 12:29 PM, Steve Edwards wrote:
On Wed, 11 May 2011, Eric Wieling wrote:
Generally you should insert a Noop in the dialplan to examine variables.
Noop(EXTEN is ${EXTEN}) for example.
The 'verbose()' application would be an example of 'better practices.'
It's function is
What I do is when ever a call comes in I update a table in MySQL to active =
(active +1). On hang up I do active = (active -1).
I have a cron that checks once a minute to see how many active and stores it
along with epoch in db.
I then have a graph that shows channel usage. If you want the
On Thu, 12 May 2011, Dovid Bender wrote:
What I do is when ever a call comes in I update a table in MySQL to
active = (active +1). On hang up I do active = (active -1).
I have a cron that checks once a minute to see how many active and
stores it along with epoch in db.
I then have a graph
Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd
like to take a look at it for sure. The dial plan example Leif replied with
is pretty much what I was thinking, just didn't have a clue how to go about
it. ;)
Haven't figured out how I'm going to display the usage
On Wed, 11 May 2011 01:09:16 +0800, Scott Zhang
macromars...@gmail.com wrote:
So does this mean no solution when used ZAP/DAHDI with PSTN line?
If I installed an E1, will that work?
Before getting an E1, maybe ISDN provides call supervision?
--
Hello,
We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now
experiencing higher CPU utilization on their server. I can't see anything
wrong, so is this just expected with 1.6? Can anyone help explain it?
Thanks for any advice.
--
David Cunningham, Voisonics
Hello Folks,
Download links on the website have not been updated (asterisk.org)
Regards,
--
Jose P. Espinal
http://www.eslackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs
--
_
-- Bandwidth and
Dear, I have Asterisk 1.6 with an E1 Digium card with echo
cancellation module. So I need to use just the echo cancellation by
hardware and disable the echo cancellation by software. I use DAHDI
for my telephony hardware.
If the lines involved with echo cancel are:
In /etc/dahdi/system.conf:
Le 2011-05-09 09:31, Jim Dickenson a écrit :
Make sure the firmware on the card is latest. I had a problem, not like your,
and flashing the card to the latest firmware resolved it.
It appears it did not change anything...
So, to re-cap, I have a sangoma A101 card, with the firmware uptodate,
In asterisk CLI do pri show spans. The fact the card is in RED alert means
the hardware does not see the pri line connected to the card.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 11, 2011, at 6:55 PM, Nicolas Ross wrote:
Le 2011-05-09 09:31, Jim Dickenson
On Wed, May 11, 2011 at 10:54:06PM -0300, Alejandro Cabrera Obed wrote:
Dear, I have Asterisk 1.6 with an E1 Digium card with echo
cancellation module. So I need to use just the echo cancellation by
hardware and disable the echo cancellation by software. I use DAHDI
for my telephony hardware.
I just reread your original email, and I think I answered the wrong
question.
On Wed, May 11, 2011 at 11:36:04PM -0500, Shaun Ruffell wrote:
On Wed, May 11, 2011 at 10:54:06PM -0300, Alejandro Cabrera Obed wrote:
Dear, I have Asterisk 1.6 with an E1 Digium card with echo
cancellation module.
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