Darrick Hartman (lists) wrote:
[EMAIL PROTECTED] wrote:
Hi John,
Try ...
carriers.icall.com - No minimum, unlimited concurrent calls, great
price, some areas US 0,009. Only USA
voipjet.com
teliax.com - Not so cheap, and they do one-minute rounding ... not good
at all. But they
Martin Smith wrote:
Hey folks,
So I'm trying to get Festival() working on 1.2.17. I'm trying to use
app_festival:
Here's the show dialplan output from that extension:
'3378' = 1. Answer()
[pbx_config]
2. Festival(Hello Asterisk caller. How is your day?)
Matt wrote:
If you have been affected by the SunRocket / ALLO folding issue,
ChiliTech would like to extend our hand to you to help you in this
time. We will transfer your numbers to us for no cost, and will
match your SunRocket or ALLO rate. Please contact us at
1-866-678-6858 x 126 or
Damon Estep wrote:
Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is
probably being send before the display IE arrives. The display IE is used for
CNAM delivery, and should not exceed 15 characters.
It is very common to put a message in the display IE that indicates
I am submitting a patch to the Bug tracker next week that will have a manager
event fired alongside every queue log write. You can then send the queue
information to the database in realtime if you have a manager interface script.
If anyone is willing to test this patch once posted, I would
Matt wrote:
I'll take either
Actually now that I have had a chance to think about what I did (sorry
bad week here). Yes, I will admit I did patrionize the users list...
sorry if I offended anyone. I just figured I'd try to help any
SunRocket users out that may not be on the biz list.
John Meksavan wrote:
The DTMF tones are being sent twice. On SIP Peer side, I set the
DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and
AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side
to use
INBAND?
From: Alex Balashov [EMAIL PROTECTED]
Mike wrote:
Thanks. Tell me, how intensive is it to use qualify? What type of
packet/check is done with this? Is it reasonnable to use qualify for
thousands of devices?
Once the device is considered to be unreachable for any number of
reasons, will another poll of the device be done
James FitzGibbon wrote:
On 8/1/07, *Linux Lover* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
This SOHO PBX box won't interop with Asterisk
because it doesn't speak any
of the protocols that Asterisk does. This box
I tend agree with your evaluation. Still, I was
Benjamin Jacob wrote:
Ouch.
And I thought I had an answer to my query.
I totaly agree abt the long disclaimer nonsense Schmaltz, but I swear by
the powers up there, it's the admins over here at my workplace doing all
that nonsensical magic, as the mails go out. I wish i had the freedom to
Forums wrote:
You may want to consider the multi-tenant version of Thirdlane's PBX
Manager (www.thirdlane.com).
I've been using for a long time and very happy with both single and
multi-tenant versions.
Benjamin Jacob wrote:
Anthony Francis wrote:
Hello good ppl,
A couple
Haudy Kazemi wrote:
On Aug 2 2007, John Meksavan wrote:
Asterisk Users,
I recently ran into some problems with the quality of service with
Teliax.
This occurred on August 1, 2007 with a dropped outbound call, audio
quality isse on the callee side- not hearing me well on callee
You know the problem is that most consumers think that it is possible to get
the best and the most reliable for almost nothing.
They go out with this expectation and get the cheapest, then when it bites them
a few times, they scream why me.
-- Original Message
Michiel van Baak wrote:
On 05:27, Fri 03 Aug 07, bilal ghayyad wrote:
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
WaitExten is waiting for
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of failure
is unreasonable.
It's impossible. I can't think of a single example where this
actually exists.
Getting even close is hideously expensive.
Erik Anderson wrote:
On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
Call Sangoma and give them root if you can. They will fix it quickly or
at least give you ammunition that it is the telco's issue.
Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal
Olivier wrote:
Thanks.
In fact, my questions are more about usage than about technical
background.
For instance, I doubt a user will log his system off when leaving :
some don't even turn their PC off.
Does anyone has an experience to share about that ?
Darryl Dunkin wrote:
wanpipemon is the way to do it as far as I know.
For starters, what do your zaptel/zapata configs look like?
I would first verify that your D-channel is set properly, you can view
that in the console as follows:
asterisk pri show span 1/0
Primary D-channel: 24
Status:
Erik Anderson wrote:
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
also in asterisk do:
pri intense debug span 1
Then you should see UA's and SABME's, If you don't, your not talking to
them.
I see plenty of SABMEs, but nothing else:
[ 02 01 7f ]
Unnumbered
Steve Totaro wrote:
Anthony Francis wrote:
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of
failure
is unreasonable.
It's impossible. I
Mark Coccimiglio wrote:
Steve Totaro wrote:
What if a train derails and slices through the main fiber connections.
OK, so you have XO, Global Crossing, Verizon, and UCN all for
redundancy. Well guess what? They are all most likely running over
those strands of fiber. You better
Mike wrote:
Hi,
Is it possible to write a function in Asterisk, that returns a value?
Sort of like any programming language allows?
For example, I`d like function ReturnSipReg to return the right
SipRegistration to dial, based on some value so that I could use it in
my dial plan:
You are looking for the AGI: http://www.voip-info.org/wiki-Asterisk+AGI
Anthony
James FitzGibbon wrote:
On 8/8/07, *Mike* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I'd be most thankful for some link to a page that shows how to
write such a
function in Asterisk.
There is
Matt wrote:
I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I
have three agents. We'll call them 101, 102, and 103.
When a call comes in.. I want it to always try 101 if no answer try
102.. if no answer try 103, etc.
However, what it is doing is... it will ring 101...
,
something obvious I missed?
Thank you,
Mike
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Anthony Francis
*Sent:* Wednesday, August 08, 2007 12:39
*To:* Asterisk Users Mailing List - Non
Jeng Yu wrote:
Hi All,
Can I overlap Playback() with Dial() in a dialplan?
For example, I have this scenario: A call comes in, Asterisk picks it up,
does Background(enter_number), then does Playback(bulletin_message),
and while the Playback() is still going, I want to execute Dial() to
Rilawich Ango wrote:
Hi all,
From the web, I can find a table scheme of sipusers for ARA using.
However, I can't find any meaning of each field, especially for the
field regserver which is new in the table. Can any tell me more
detail about the usage of each field?
CREATE TABLE
Mike wrote:
Ok, now that I've learned I cannot use any variables when using the
`hint` priority (for BLF), I figured I'd try to use the next best
thing: hardcoded values using realtime. This way I avoid variables
such as ${ACCOUNTCODE} but I can at least change the DB more easily
than
PROTECTED]] On Behalf Of Anthony
Francis
Sent: Thursday, August 09, 2007 12:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The quest for making hint more
flexible
continues - using Realtime now
Mike wrote:
Ok, now
Andres Paglayan wrote:
Hi,
is it possible to fork from a dial plan?
meaning, is there a way to redirect to two different
context,extension,priority
without waiting for the first to finish?
Andres Paglayan
--Harmony is more important than being right
Bapak
Andres Paglayan wrote:
Hi,
is it possible to fork from a dial plan?
meaning, is there a way to redirect to two different
context,extension,priority
without waiting for the first to finish?
Andres Paglayan
--Harmony is more important than being right
Bapak
Mike wrote:
The thing is that I make them automagically reload from outside Asterisk (by
calling asterisk -rx extensions reload)
Mike
Why would you do that? There is no real point in reloading configs
unless they have changed.
Anthony
___
Mike wrote:
Well, if you really must know (this is OT for everybody else I guess) I have
a custom Web GUI used for my customers, and when some settings are modified,
a conf file is created. This conf file must be reloaded at this point,
therefore I call the reload command externally.
Why do
Gordon Henderson wrote:
On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it.
I can't help but think you're making life hard for yourself.
Why not do it by
Todd Adamson wrote:
As I am working my way to understand Asterisk, I have a couple of
questions that hopefully someone will answer.
I know there are differences between a PBX switch and a CO switch.
Can Asterisk completely replace and act as a CO switch? Are there any
telecoms out there
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really? http://www.pt.com/products/prod_segway_ntwksolution.html
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Mike wrote:
Ken,
You understood correctly.
For those who answered and didn't understand the need, I wanted to
reload automatically (based on some external event) part of my conf
file. I only felt that since this was automatic, it would have been
better to limit this reloading to
First, this is a non-commercial list, please do not post that stuff here.
Alex, cdr csv is less then efficient for reporting, you should drive
your cdr's to a database and then you can do some good reports based on
that.
Anthony
Alex Balashov wrote:
We at Evariste have a lot of experience
Adrian Marsh wrote:
Hi All,
I was made aware today that some of my calls coming in are not going to
voicemail... Below are some logs, and the macro that should run on the
incoming_pstn context for that extension. I can see that theres a
non-zero exit before it gets to voicemail, but I've
dialed the number
incorrectly.. But that's no-where in the dialplan, and I do see the
incoming calls correctly for the times he's saying..
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: 14 August 2007 15:40
looks broken, is there an apps dir in the source directory?
Mark Quitoriano wrote:
is there a new way to install asterisk? im using centos 4.5 and trying
to install asterisk. when i do make clean and make install i get this
error.
# make clean
--snip--
make[1]: Leaving directory
You want a key system, the fianl frontier of an asterisk implementation,
and currently my holy grail.
The best way to do it in an ugly way is to park the call and have a
speed dial for pickup. Some phones like Aastra 55i and 57i can even have
their hold button reprogrammed to blind transfer to
Since I dont use 1.4 then you tell me. :)
Stephen Bosch wrote:
Anthony Francis wrote:
You want a key system, the fianl frontier of an asterisk implementation,
and currently my holy grail.
The best way to do it in an ugly way is to park the call and have a
speed dial for pickup. Some
Stephen Bosch wrote:
Anthony Francis wrote:
Since I dont use 1.4 then you tell me. :)
This functionality is supposed to be supported in 1.4, though I've never
personally tested it. When it's configured it gives the key system
behaviour you describe.
-Stephen
http://www.faqs.org/rfcs/rfc3261.html
Rizwan Hisham wrote:
Hi All,
Can anybody send me a complete list of sip events. i know only 3 of
those whihc are register, message-summary, message notification.
message-summary event is causing some problems actually. My client
sends a bad-event
You will need to extend your schema to include all of the attributes
that can be used in sip.conf plus the extra ones that allow realtime to
store connection information. Please refer to the realtime info at
voipinfo.org to get a feel for what your schema should look like.
Anthony
Abhishek M
You are a victim of hung channels, just write a script that corrects this.
Anthony
Mail list wrote:
The destination numbers are valid in almost all cases . But i do think
it might be when someone is on call and on client side internet
connection goes off .. I am really not sure about this
Bill Andersen wrote:
Gordon Henderson wrote:
Out of curiosity, what's the GUI you are currently using and what do you
feel are it's limitations?
It is a commercial product called Evolution PBX
by Intuitive Voice Technology (IVT). I don't want to imply
I'm unhappy with it,
Get the providers support on the line and I will bet anything you are
not hitting their side of the line, this is most likely a signaling or
channel numbering issue.
Anthony
Veselin Kantsev wrote:
Hello,
I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some
Snom 300 and
Peder @ NetworkOblivion wrote:
Does anybody have realtime queue members working? Not the queues
themselves, just the members. I have realtime working for voicemail and
sippeers, but I can't get queue members to work. Here is what I have:
res_mysql.conf:
[general]
dbhost = 127.0.0.1
Peder @ NetworkOblivion wrote:
Anthony Francis wrote:
There is no queue_members file, asterisk doesnt know hat you are
talking
about, you would have to #include queue_members from inside that queue
definition.
Huh? How is including a file going to make realtime access
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Stephen Bosch wrote:
Anthony Francis wrote:
dial(SIP/polycom-on-my-deskLocal/5551212,15,tr)
Will this work even if the Local is pointing to a Zap channel?
As far as I know, this only works with SIP or IAX outgoing.
-Stephen-
I use local because It dials the call from default
I concur, Centos 4.4 FTW. ^^
-- Original Message --
From: Edgar Guadamuz [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Date: Fri, 24 Aug 2007 23:50:51 -0600
I have used CentOS and it
:
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Rockynet VOIP
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[EMAIL PROTECTED]
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asterisk
to translate that yourself on your
own side. If all you did was match a _011X, you might never GET it
unless someone truly dialed a 011
MOST of our phones send a + as a +, and we see it often.
N.
Anthony Francis wrote:
To match any single digit use X. Also, it is simplest
Adrian Marsh wrote:
Hi Dovid,
Because there may be complex logic in other parts of the context for
handling different countries in a different way, so I wouldn't want to
duplicate that Dial logic. Easier to jump back to the beginning of the
context and have the digits replaced.
I original
SIP wrote:
(many of our users do that, and they just type a + like a normal human)
I don't know if you intended to be rude with the normal human comment but it
sure seems like it when reading your reply. Also how many users know they can
dial ** to get a +? Especially when so many cannot as
I knew that was true about GSM networks outside of the US, but to be
honest, I am not concerned with those networks ^^.
Mr Shunz wrote:
On 8/31/07, Anthony Francis [EMAIL PROTECTED] wrote:
I don't know if you intended to be rude with the normal
human comment but it sure seems like it when
So yeah, I can admit that this is all true outside the US. ^^, sorry.
Steve Kennedy wrote:
On Fri, Aug 31, 2007 at 10:03:07AM -0600, Kai-Uwe Jensen wrote:
On 8/31/07, Anthony Francis [EMAIL PROTECTED] wrote:
Mindfully wanting to use a + instead of knowing the international access
Not being concerned does not == ignorant.4
Benny Amorsen wrote:
AF == Anthony Francis [EMAIL PROTECTED] writes:
AF I knew that was true about GSM networks outside of the US, but to
AF be honest, I am not concerned with those networks ^^.
On 8/31/07, Anthony Francis [EMAIL
://lists.digium.com/mailman/listinfo/asterisk-users
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Ove Aursand wrote:
Abdul wrote:
Hi expets,
I have installed Asterisk 1.4.11 on CentOS4 successfully without any
error.
But when i am trying to start asterisk with following cmd i am
getting unknown command.
[EMAIL PROTECTED] ~]$ asterisk -vvc
-bash: asterisk: command not found
Ove Aursand wrote:
Anthony Francis wrote:
Ove Aursand wrote:
Abdul wrote:
Hi expets,
I have installed Asterisk 1.4.11 on CentOS4 successfully without any
error.
But when i am trying to start asterisk with following cmd i am
getting unknown command.
[EMAIL PROTECTED
Awhile back I had heard some talk, in this list I believe that Agent
callback login was going to be deprecated in 1.4, I see it is still
there. Does anyone know what is happening with this?
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[EMAIL
the Adtran relationship goes way back...
Thanks,
Steve Totaro
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Rockynet VOIP
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Not only that but AEL doesnt mesh with realtime.
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Rockynet VOIP
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[EMAIL PROTECTED]
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Kevin P. Fleming wrote:
Anthony Francis wrote:
So the most helpful thing would be a solid example of how to exactly
duplicate the agent callback login behavior in a real-time friendly
manner. The part I am missing is how we are to do authentication.
Please define what you mean
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Kevin P. Fleming wrote:
Anthony Francis wrote:
Right I am just saying that I can't use AEL in the DB.
My dialplan is in the DB, not the agents.
It shouldn't be that hard to translate the AEL example into traditional
dialplan language; in fact, Asterisk does that itself when you
Al lists wrote:
I'm using Linksys Wip300 and i'm not happy with it.
On 9/13/07, *Dave Walker* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
Hi folks:
I know it's come up a few times before, but I need some
thanks to XXX.XXX.XXX.XXX or
something along those lines.
This helps you know what is in SIP messages:
http://www.ietf.org/rfc/rfc3263.txt
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first stop when trying to do
basic asterisk things.
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was either not supported (1.2) or not viable (GUIs,
realtime, resistance to change, etc.).
--
j.
Thank you for the awesome help!
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Rockynet VOIP
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Jared Smith wrote:
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote:
. matches any number of the preceding character, change it to _X.*X.
That still won't help. Once the Asterisk pattern matching parser sees a
period in the pattern, it ignores anything after it. (I'm
that you don't think you got
enough service from volunteers is a bit preposterous.
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Easy solution == pay by performance.
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asterisk
.
Sounds like both you and bkw know what the difference is but don't
really know how to explain it...
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Alex Balashov wrote:
So, is a pure VoIP switch by definition not a softswitch, despite whatever
other characteristics it might have?
On Wed, 19 Sep 2007, Anthony Francis wrote:
A real softswitch uses TDM
(http://en.wikipedia.org/wiki/Time-division_multiplexing) and Asterisk
uses
: unreachable, not registered
8: ringing
I've recently seen 16 (== hold?) but can't find that value
documented anywhere.
Regards,
Philipp Kempgen
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,
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Philipp Kempgen wrote:
Anthony Francis wrote:
Here is what I use.
sub devstate2str($)
{
#func name stolen directly from asterisk
#takes int devstate and returns string val
my $ids = shift;
my $devstatestring = {};
$devstatestring-{0} = Unknown; #0
Philipp Kempgen wrote:
Anthony Francis wrote:
Oh one other note, when asking questions such as this, it is really wise
to include which version # you are using.
Right. Sorry.
1.4.11 (for the archives)
Regards,
Philipp Kempgen
That is what I thought, makes what I said
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Rilawich Ango wrote:
What do you mean? I just want to know whether there is a way to do
the following.
1. A --calls -- B
2. A on hold, B --calls -- C
3. A, B and C connected to talk
On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote:
How are you going to do it without a phone?
PaulH
Tomás Laureano Peralta Tormey wrote:
Brian:
Maybe the CLI command stop gracefully is what are you looking for.
Basically, Asterisk will stop receiving incoming calls (of any channel
type) and stop itself when all the current calls finish.
I hope this help you.
Best regards, Tomás.
rendering you unable to utilize realtime.
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Thank you and have a wonderful day,
Anthony Francis
Rockynet VOIP
(303) 444
out loud.
Regards,
Philipp Kempgen
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Thank you and have a wonderful day,
Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]
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asterisk-users mailing list
Anselm Martin Hoffmeister wrote:
The problem there is that you have a very small windows. AFAIK there
are no tftp servers that can generate files on-the-fly, so your script
You could make a perl script that pretends to be a TFTP server. Then it
could generate the file on the
john beaman wrote:
For pinout info, check out: http://www.asteriskdocs.org/cables/
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 10/26/2007 4:01:29 PM
Michelle Dupuis wrote:
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