Hello,
I am trying to make a redial script. The script i tried is as follows.
The problem with the script is, there are plenty of dialing plans on
asterisk for different users, only internal, local international etc..
As you see i wrote as dlpn_local temporarily, because, i couldnt use
Dial() comma
On Thu, Dec 24, 2009 at 12:13:55PM +0530, Goyal, Amit wrote:
> Hi All,
>
> Can some help me with how to run Asterisk with gdb.
What specifically do you want to do? What do you want to check?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-795240
Hi,
How would I go about troubleshooting this:
[Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum
retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for
seqno 101 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 re
Hi All,
Can some help me with how to run Asterisk with gdb.
Thanks,
Amit
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On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham wrote:
> AsteriskWin32 does have SIP server functionality, same as the linux
> version.
>
> I can't think of any reason why having your CentOS Asterisk be both client
> and server and register with itself wouldn't work.
> Although I am wondering h
Hi,
Is it possible to create a meet me room on the go through dial plan? I am
looking to use AMI Originate to drop a call into meetme room and once it's
proved that party is joined, play him an announcement, grab few numbers from
them, and then dial a second number and drop into the same meetme r
Hi,
how can I get agent status (not in use,busy,ringing,etc) from my
dialplan?I want to collect all available agent before entering a
queue.Please help me.
Thanks a lot.
Best Regards,
Francis
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Hi list,
We have a client that wants to hook up their SIP trunks to their
existing NICE (seattle based company) analog phone recorder. Whenever
they make a phone call using their SIP provider, they want this
conversation to be played out to NICE recorder.
I currently have a hack using an FXS for
hi,all
i have load cdr_radius.so successfully! but another error occur.
-- Executing [4...@tutorial:1] Dial("SIP/ivan-0a07dc80",
"SIP/test") in new stack
-- Called test
-- SIP/test-0a08b0f0 is ringing
-- SIP/test-0a08b0f0 answered SIP/ivan-0a07dc80
-- Packet2Packet bridging SIP/ivan-0a0
Thank you !
i have load cdr_radius.so successfully! but another error occur.
-- Executing [4...@tutorial:1] Dial("SIP/ivan-0a07dc80",
"SIP/test") in new stack
-- Called test
-- SIP/test-0a08b0f0 is ringing
-- SIP/test-0a08b0f0 answered SIP/ivan-0a07dc80
-- Packet2Packet bridgi
Kevin P. Fleming wrote:
> Barry Fawthrop wrote:
>
>> I have been looking around and haven not been able to find a working example
>> I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
>> 1.4.10.2
>>
>> I use a sangoma A200 card so I am using wanpipe 3.4.7
>> If I use zap
Has anyone experienced this problem before?
I am running Asterisk 1.4.21.2
If I run:
MixMonitor(..)
Dial(SIP/...)
Both parties cannot hear each other.
As soon as I comment out MixMonitor, the audio can be heard.
I saw this issue on https://issues.asterisk.org/view.php?id=16256
It seems to match
Amazing. Thanks Jarrod.
On Wed, Dec 23, 2009 at 7:09 PM, Jarrod Lash wrote:
> Bruce,
>
> http://www.voip-info.org/wiki/view/Asterisk+manager+API
>
> In the AMI do :
> Variable: crap1=data
> Variable: crap2=data
> Variable: crap3=data
>
>
> In the dialplan variables will be in the format..
> ${c
Jarrod,
Thanks for the input. Can you please include a sample of your work? It will
really save me days of headache and tests if I can start with something that
is tested to work.
I really appreciate your response.
In the meantime, I will go check meetme creation rules.
Regards,
Bruce
On Wed,
Bruce,
What I have done for apps like this is call the first guy and at the
end of your dialplan put him in a meetme room. In your script watch
for the meetme room to be created in the AMI output.
Once the room is created originate a call to the other guy and dump
him into that meetme room when
Bruce,
http://www.voip-info.org/wiki/view/Asterisk+manager+API
In the AMI do :
Variable: crap1=data
Variable: crap2=data
Variable: crap3=data
In the dialplan variables will be in the format..
${crap1}
Also as an FYI.. Look at cepstral in place of Festival...
--
Jarrod Lash,
Federated Commu
Hello Everyone,
I want to send a few numbers (variables) when doing Asterisk AMI Originate
command and then have Festival read them back to customer in the context.
How should this be done? Following is my not working example and also
reference on AMI Originate Command:
Command: Originate
Channel
Hi Guys,
I am trying to make a web form where a person is allowed to put in
$phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller
ID. There are a few problems that I am facing with Asterisk AMI Originate
command. The reason why I want to use the darn AMI Originate is because
I would recommend the 2U R710 as it has 4 PCI Express slots which
should provide plenty of expansion capability. I have a Dell R710 in
the testing phase installed with 2 Sangoma A104E cards and a Sangoma
A102E card. Everything has been working great and the hardware is
solid. It is going to replace
On 12/23/2009 03:48 PM, Fred Posner wrote:
> On Dec 23, 2009, at 4:21 PM, Sascha Ferley wrote:
>
>> Hi,
>>
>> I am in need of ordering a new server here for our asterisk solution. Since
>> the corporate standard is Dell we need to stick to a dell server. We used to
>> deploy 2900III without any iss
Hi guys,
My Asterisk box is connected to two different SIP providers. I have one
interesting problem with both of them: from time to time, SIP provider would
not send the BYE command, eventhough the person on the other side already hung
up the call. So the line gets stuck active until someone o
On Dec 23, 2009, at 4:21 PM, Sascha Ferley wrote:
> Hi,
>
> I am in need of ordering a new server here for our asterisk solution. Since
> the corporate standard is Dell we need to stick to a dell server. We used to
> deploy 2900III without any issues, however now they are almost not available
>
Oops !
The sendmail macro was missing, sorry !
[macro-Sendmail]
;===
; ARG1 = Address To
; ARG2 = Address From
; ARG3 = File attachment
; ARG4 = Pages Qty
; ARG5 = Rate
; ARG6 = HeaderInfo
; ARG7 = Remote
Sascha Ferley wrote:
> The biggest issue I can see is that in the future we may want to get a
> transcoder card, however none of the new servers have a standard PCI slot
> available any more as with the new Nathelem chips having gotten rid of the
> basic bridge I guess.
http://www.digium.com/en/p
Sascha Ferley wrote:
> Hi,
>
> I am in need of ordering a new server here for our asterisk solution. Since
> the corporate standard is Dell we need to stick to a dell server. We used to
> deploy 2900III without any issues, however now they are almost not available
> any more and are looking at a
Hi Francois,
here is Francois too ;-)
Check that :
[fax-outbound-calls]
exten => _X.,1,Dial(${ACROPOLIS}/${EXTEN},,G(fax-tx^send^1))
[fax-tx]
exten => send,1,NoOp( SENDING FAX )
exten => send,n,Set(FaxTxDir=/var/spool/fax/tx/)
exten => send,n,Set(FAXFILEPDF=fax-${FAXCOUNT}-tx.pdf)
exten
Hi,
I am in need of ordering a new server here for our asterisk solution. Since
the corporate standard is Dell we need to stick to a dell server. We used to
deploy 2900III without any issues, however now they are almost not available
any more and are looking at a new solution.
Has anyone tried an
On Wed, Dec 23, 2009 at 10:49 AM, BERGANZ François
wrote:
> Hello,
>
>
>
> I need to send a tiff via fax with my asterisk 1.6.1.0.
>
> I tried in the dialplan
>
>
>
> [default]
>
> exten => _X.,1,SendFax(/root/test.tiff)
try "originate sip/provider/number extension 1...@default"
Martin
Hi
I use one of these http://www.soekris.com/net5501.htm fairly cheap to
buy and to run, I have a tdm410 in there and it has worked flawlessly
I am running debian i386 on the box - it also doubles as my
firewall/router/vpn/adsl box.
I do have one problem with the box (but I have seen on other bo
Thanks very much Kevin
But its not that clear - in fact the support email address isn't listed on the
support page - everything leads to logging in with your registered product. Its
incredibly frustrating and I recommend you try looking and seeing how it works
for yourself if you haven't got yo
Duncan Turnbull wrote:
> I did get one response which was to email customer services and eventually
> found an email address for them but that seems to have fallen on deaf ears.
> Perhaps my expectations are too high but it was an email a week ago and no
> response, not even to say go away.
Th
Hi there
I have a client who has an AA50 from DIgium. I am really challenged getting any
support as the client doesn't have any of the original registration or
subscription info, someone did the install and left without any records. I
thought okay we can ask Digium, but you can't get help wthou
I have asterisk 1.6.1.10 and a Rhino CB24 Channel Bank...
A few channels seem to have locked up... If I plug an analog phone in
the port, I get either dead air or a busy tone...
Is there any way to reset this channel without restarting asterisk?
___
"Sip show users" or "sip show peers" should do the trick, but I'm not sure
about 1.2; all of my experience is in the 1.4 branch.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Wednesday, December 2
Hi,
This is the first time I experience this problem with Asterisk:
all of a sudden SIP registrations stopped working. Active calls kept working
but new calls could not be established (I did NOT perform a "graceful
restart").
Besides, would a "restart gracefully" actually deny SIP registration
For what it's worth I'm also using HylaFAX 6.0.3 with IAXmodem to talk to my
1.6.0.17 box to send faxes across my SIP provider (they only support SIP at
this time). I can't really speak to the reliability for high-volume loads, but
I haven't had any problems with the dozen or so that I've sent o
Someone posted a message suggesting someone try sendtext() and so I
thought I'd see if it was useful. Much searching through help at the
CLI has failed to find any help for sendtext, but I did find that:
"core show function vmcount" fails but:
"core show function VMCOUNT" works.
Is that a bug
BERGANZ François wrote:
> I need to send a tiff via fax with my asterisk 1.6.1.0.
>
> I tried in the dialplan
>
>
>
> [default]
>
> exten => _X.,1,SendFax(/root/test.tiff)
>
>
>
>
>
> but I have:
>
> salledeconf1*CLI> console dial 1...@default
You are not going to be able to use Sen
Hello list,
I'm having problems placing the 2nd call via my provider. The first call
goes through and I can talk normally, but when I place the second call,
it doesn't go through and the first call is disconnected. The connection
is 20mbps downstream and 1mbps upstream, so bandwidth is not an issu
The problem isnt in my tiff images, I could use it with a paton.
I still stryed to stand on one foot offcourse J
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny
Nicholas
Envoyé : mercredi 23 décembre 2009 17:58
À : 'Asterisk
Did you stand on one foot and hold out your tongue when you made the tiff??
:-) The tiff has to be a very specific format
I spent days making my
output tiffs match the format of a received tiff I was able to send.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-u
Hello,
I need to send a tiff via fax with my asterisk 1.6.1.0.
I tried in the dialplan
[default]
exten => _X.,1,SendFax(/root/test.tiff)
but I have:
salledeconf1*CLI> console dial 1...@default
[Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to
re-open DSP devi
Magnus Benngård wrote:
> Is it in the "trunk" version or will it be added there?
It's trivial to look for yourself:
http://svn.digium.com/view/asterisk/trunk/CHANGES?view=co
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Is it in the "trunk" version or will it be added there?
On Tue, 22 Dec 2009 08:12:40 -0600, "Kevin P. Fleming" wrote:
Magnus Benngård wrote:
> Is it possible, when placing a call that u see the name of the extension
> in your diplay?
>
> For example, 2 sip.conf entries:
> [971]
> callerid="S
I'm using hylafax/aixmodem for a fax solution. If your SIP DID provider has a
T.38 path to you, faxing should be pretty reliable. I'm on * 1.4 and I'm not
too familiar with current fax offerings in 1.6. Hylafax can be configured to
email incoming faxes as a PDF, or to accept outbound faxes from
- "Barry Fawthrop" wrote:
> I have a teliax provided SIP phone number which will be the fax number
> to receive all faxes
> and have them emailed to a central email address, hopefully in PDF
> format. where they can
> be printed and/or forwarded.
>
Faxing over VoIP is not likely to work. Goo
Because we only actually read a percentage of the threads (since there are a
couple thousand per month)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Wednesday, December 23, 2009 9:01 AM
To:
Barry Fawthrop wrote:
> I have been looking around and haven not been able to find a working example
> I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
> 1.4.10.2
>
> I use a sangoma A200 card so I am using wanpipe 3.4.7
> If I use zaptel which I read I need for app_r
I apologize for my poor English.
So, i don't really understand 'how to' realize thus
When you use the cmd Dial and want to get $ from caller channel to callee (or
callee channel from caller), which way is the right way ?
Sorry, i've take a look to the wiki and asterisk code and is'nt lim
On 23 Dec 2009, at 14:48, Danny Nicholas wrote:
> Why are there three branches of 1.6?
Why are there a million threads asking this question?
;)
S
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To
On Dec 23, 2009, at 9:48 AM, Danny Nicholas wrote:
> Why are there three branches of 1.6?
>
http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/
There's more info on the blogs about the new method and long term releases.
__
On Wed, Dec 23, 2009 at 08:48:34AM -0600, Danny Nicholas wrote:
> Why are there three branches of 1.6?
There are three different releases, 1.6.0, 1.6.1 and 1.6.2 .
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.c
Why are there three branches of 1.6?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Wednesday, December 23, 2009 8:43 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sho
On Wed, Dec 23, 2009 at 08:22:28AM -0600, Danny Nicholas wrote:
> Pardon my ignorance, but wouldn't 1.8 be the start of a whole new branch of
> Asterisk (1.0, 1.2, 1.4, 1.6, 1.8)? What is the projected timetable for
> rc/release on 1.8?
You're almost correct. 1.8 is a whole new branch of develope
Hi All
I have been looking around and haven not been able to find a working example
I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
1.4.10.2
I use a sangoma A200 card so I am using wanpipe 3.4.7
If I use zaptel which I read I need for app_rxfax then asterisk crashes
Pardon my ignorance, but wouldn't 1.8 be the start of a whole new branch of
Asterisk (1.0, 1.2, 1.4, 1.6, 1.8)? What is the projected timetable for
rc/release on 1.8?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behal
This is sort of counter-intuitive, but would accomplish the OPs goal; when
calling an extension, instead of answering, have the other line call you
back; this would take 1-5 seconds to accomplish.
- exten => 100,1,noop(call joe)
- exten => 100,n,agi(callback.agi|${callerid(nu
This may work on 1.6, but on 1.4(26.3), it just leaves an Instant Message;
not the desired result for OP.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alec Davis
Sent: Wednesday, December 23, 2009 2:23 AM
To: 'Asterisk Us
Hi,
I am running a Asterisk 1.6.1.6 (soon to be upgraded) PBX for a client and they
are having a issue that they are unable to reach a TFN (Toll Free Number).
When they call a automated announcement is received that the number will not
accept calls from the originating area code.
It has been
AsteriskWin32 does have SIP server functionality, same as the linux version.
I can't think of any reason why having your CentOS Asterisk be both client
and server and register with itself wouldn't work.
Although I am wondering how much help all this will be in debugging a
connection problem to ano
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham <
dcunning...@voisonics.com> wrote:
> Hadi,
>
> You could use Asterisk as a sip server, it's installable on Windows.
>
> Using "sip set debug on" might help you with the "Host '192.168.0.139' does
> not implement 'REGISTER'" problem.
>
>
> On Wed,
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham <
dcunning...@voisonics.com> wrote:
> Hadi,
>
> You could use Asterisk as a sip server, it's installable on Windows.
>
> Using "sip set debug on" might help you with the "Host '192.168.0.139' does
> not implement 'REGISTER'" problem.
>
>
> On Wed,
Wednesday, December 23, 2009, 11:17:39 AM, ABBAS wrote:
>> when compiling asterisk with Postgresql we need to specify directory where
>> the postgresql is installed.
>> I need to know once asterisk is ready to use(ie compiled and installed ).
>> Do it still refer the postgresql files that are not
23 dec 2009 kl. 11.25 skrev David Cunningham:
> Shukun,
>
> It tells you "No such file or directory". Is the file in your modules
> directory?
Actually, to be more specific. The module cdr_radius.so exists, but can't bind
to the radius library "libradiusclient-ng.so.2".
Check LD_LIBRARY_PATH
Shukun,
It tells you "No such file or directory". Is the file in your modules
directory?
On Wed, Dec 23, 2009 at 10:09 AM, Zhang Shukun wrote:
> hi , all
> when i do the command "module load cdr_radius.so" ,error happens.
> i have installed radiusclient-ng , what's wrong with it? thanks!
>
Jonas,
Some possible causes:
- File permission problem
- Firewall blocking
- Other network problem like no route
On Wed, Dec 23, 2009 at 10:20 AM, jonas kellens wrote:
> Calling my home numbers has always worked. Till now. The Asterisk CLI show
> the following :
>
> [Dec 23 10:53:22] NOTICE[251
Hadi,
You could use Asterisk as a sip server, it's installable on Windows.
Using "sip set debug on" might help you with the "Host '192.168.0.139' does
not implement 'REGISTER'" problem.
On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote:
>
>
> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wr
Calling my home numbers has always worked. Till now. The Asterisk CLI
show the following :
[Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite:
Failed to authenticate on INVITE to ';tag=as5b139383'
And after restarting Asterisk, the CLI is flooded by :
[Dec 23 11:11:06
Can some body shed some light on this please
On Mon, Dec 21, 2009 at 6:41 PM, ABBAS SHAKEEL
wrote:
> Hello
>
> when compiling asterisk with Postgresql we need to specify directory where
> the postgresql is installed.
> It uses some files from bin folder of postgresql (I am not a developer of
> as
hi , all
when i do the command "module load cdr_radius.so" ,error happens.
i have installed radiusclient-ng , what's wrong with it? thanks!
error message as follow:
ZHANGSHUKUN*CLI> module load cdr_radius.so
Unable to load module cdr_radius.so
Command 'module load cdr_radius.so' failed.
[Dec
Thank you! i have solved the problem. i have changed the resolution from
800*600 to 1024*768 in my vmware virtual machine.
2009/12/23 Olle E. Johansson :
>
> 23 dec 2009 kl. 10.16 skrev Zhang Shukun:
>
>> hi, all
>> when i run "make menuselect", it say
>>
>> Terminal must be at least 80 x 21
23 dec 2009 kl. 10.16 skrev Zhang Shukun:
> hi, all
> when i run "make menuselect", it say
>
> Terminal must be at least 80 x 21.
> menuselect changes NOT saved!
>
> in the bottom message, what's wrong?
Terminal must be at least 80x21
You need a terminal window that handles at least 80 ch
On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote:
>
> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote:
>
> >
> >
> >
> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner
> wrote:
> >
> > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote:
> >
> > > Dear All
> > > I have an application that calls f
hi, all
when i run "make menuselect", it say
Terminal must be at least 80 x 21.
menuselect changes NOT saved!
in the bottom message, what's wrong?
Thanks!
--
Regards,
Sucan
___
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On Tue, Dec 22, 2009 at 11:41 AM, Dan Journo
wrote:
> I recommend you follow the detailed install guide in this book and
> install all the required support programs etc.
>
> http://downloads.oreilly.com/books/9780596510480.pdf
>
>
>
>
> --
>
> *Thank you for contactin
23 dec 2009 kl. 08.53 skrev jonas kellens:
> Can I define the realm on a per peer basis ??
> Can I define a realm to be used for one peer and another realm for another
> peer in sip.conf ??
>
> I have an ITSP that I need to authenticate with a realm that they set. But
> this realm is not valua
You may be able to use the SendText application, Conceptually and from
memory
exten => 971, 1, Answer()
exten => 971, n, SendText(Magnus <971>)
exten => 971, n, Dial(SIP/971)
exten => 971, n,
exten => 975, 1, Answer()
exten => 975, n, SendText(Stefan <975>)
exten => 975, n, Dial(SIP/975)
e
23 dec 2009 kl. 06.17 skrev prasha...@digilink.in:
>
> Hi,
>
> How asterisk distinguish whether the re-invite is for codec change or for a
> session refresh? I know that it checks the session version and decides the
> same. But even if session version is different from the initial invite and
>
> At 5:01 PM on 22 Dec 2009, Oguzhan Kayhan wrote:
>
>> Hello,
>> Our asterisk is connected to an ericsson pbx by PRI.
>> What i want is the asterisk clients should call operator numbers by
>> dialing 0
>>
>> But, when a call is made to ericsson via number 0, it assumes that the
>> call is made f
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