Hi
this is my setup:
Customer - PRI - Server A with G729 - IAX2 Trunk(G729) - Server B
- SIP Exten allowed codec=g729 - Snom phone Agents
setup is working fine.
I want when my agents are not available (queue) like not logged in or all
are busy so no calls should come to my server b from
Josué Conti wrote:
Hi all, good? I would like to know if the option exists to together
integrate the function of queue with the voicemail of the agent, or
the pilot of the group. For example, in case that none of the agents
of queue obtains to take care of a call, this call would be directed
Tom Rymes wrote:
I dunno, I guess I'm not your mother, but then again, it seemed
pretty rude considering the guy offered the program for free and you
were criticizing the fact that he didn't develop a free linux app for
you, too.
Not specifically directed at Toms reply -
Gee, all Stephen
You could alternatively set a context for your queue in your config and
create an extension for voicemail, if you would rather give the option to go
to voice mail to the caller... (example: They can dial 0 to leave a message)
On 5/4/07, Per Jessen [EMAIL PROTECTED] wrote:
Josué Conti wrote:
Cesar Benjamin Garcia Martinez wrote:
Somebody can tell me, what way i can send/receive faxes with asterisk
1.4???
[snip]
How to i can send/receive fax to/from PSTN on asterisk 1.4 ?
Check out a very recent thread on just that subject. Or go study how to
use iaxmodem and hylafax.
/Per
Tzafrir;
Actually I have found this config to work really well. I prefer to
use a script run from inittab but Ubuntu doesn't work like Redhat or
BSD. On a production box keeping asterisk up and running is THE TOP
priority. If you would rather check every five minutes then replace the
Steve,
I didnt mean to say that your patch did that. Actually i did saw this error
before applying your patch. i just mentioned it here. So is this problem
fixable?
On 5/5/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Fri, 2007-05-04 at 15:25 +0500, Rizwan Hisham wrote:
Nops. removing
Hi
I've few queues configured in * box is there any what that before sending
call to a particular queue can we get the status of the queue that is how
many agents are available in this queue (logged in, paused, busy,
unavailable).
thanks
arun
___
Hello,
Does anybody know whether Asterisk 1.4 supports TLS? Or may be any work
patches or branches?
Thanks in advance
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Alexander Olekhnovich
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Hi Josue,
Yes you can use Asterisk along side an existing PABX.
So with your existing Avaya you can allow it to connect to the handsets, but
when calls are received for voicemail then you can send them to the Asterisk
server another functionality you might find useful are conference
Hi,
Is there any way that I can store my manager API output that is:
My question is that is there any why using that I can get the QueueStatus
and store the result in some text file for further processing.
?php
$strHost = 127.0.0.1;
$strUser =
I've reposted with a more meaningful subject - hopefully someone will
replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP.
The registration succeeds, and is confirmed with SIP SHOW REGISTER.
However, we frequently (every few minutes) see this on our console:
REGISTER
Hi all, thanks for this reply.
Follows below the current configurations of mine asterisk, where the line
functions perfectly, but does not obtain to rotear in case that no agent
takes care of, for the voicemail. How it could give an option to the caller
so that it can send a message? Sample: it
Arun Kumar wrote:
Is there any way that I can store my manager API output that is:
My question is that is there any why using that I can get the QueueStatus
and store the result in some text file for further processing.
?php
$strHost = 127.0.0.1;
Hi Dean, thank you will be this attention.
Currently asterisk is interconnected in pabx legacy through a A104D with
protocol ISDN Qsig, uses LCR for routes of lesser cost and calls for other
localities. But I see that domains of asterisk is limitless therefore would
like to use it as also
On Sat, 5 May 2007, Arun Kumar wrote:
Hi,
Is there any way that I can store my manager API output that is:
Read The Fine WiKi!!!
http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+PHP
Gordon
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Ive done a lot of work with Avaya. Voicemail systems attaché dot Avaya use
Qsig trunk to pass calls to voicemail servers. The core of their modular
messaging/message networking infrastructure can also use VPIM for
communication between vmail servers. As far as I know, you cant use
Asterisk in
Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a
precio de tarjeta usada...
saludos,
Rodrigo Mercado S.
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Mark Coccimiglio wrote:
Tzafrir;
Actually I have found this config to work really well. I prefer to
use a script run from inittab but Ubuntu doesn't work like Redhat or
BSD. On a production box keeping asterisk up and running is THE TOP
priority. If you would rather check every five
Hello -
Well I've been able to find a bit more about my problem.
Again - I am not bound to a specific interface (0.0.0.0)
When a SIP invite addressed to the .36 address, Asterisk replies FROM the
.38 address. Is this the expected behavior?
Wouldn't it make sense for Asterisk to reply on FROM
On Sat, May 05, 2007 at 06:23:43PM +0200, Remco Post wrote:
Mark Coccimiglio wrote:
Tzafrir;
Actually I have found this config to work really well. I prefer to
use a script run from inittab but Ubuntu doesn't work like Redhat or
BSD. On a production box keeping asterisk up and
On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote:
Alguien tiene una TDM400P con modulo FXS usada a la venta ??,
obviamente a precio de tarjeta usada...
saludos,
Rodrigo Mercado S.
For anyone who is not a Spanish speaker, Rodrigo is looking for a
used TDM400P card with FXS modules.
Hi,
I have some annoying telemarketer calling me on a recurring basis,
but I'd like to discourage them a bit and have some fun.
I found this:
http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture
but the link to download the sound files is dead (wyoming.e-tools.com
is
Adam Jacob Muller wrote on 5/5/07 1:06 PM:
Hi,
I have some annoying telemarketer calling me on a recurring basis, but
I'd like to discourage them a bit and have some fun.
I found this:
http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture
but the link to download the sound
You can add another line like
exten=0,1,VoiceMail([EMAIL PROTECTED])
this will catch the dialing of 0 before or after it enters the queue... but
if you want them to be able to do that while in the queue then you need to
add to your queue config a line like
context=your-queue-context
and then
Chile.
No hay listas en español, y si lo enviè en español es justamente porque si
alguien no lo habla no puede estar en CHILE, de todas formas muchas gracias
por la amabilidad de traducir mi correo.
saludos,
bye bye
On 5/5/07, Tom Rymes [EMAIL PROTECTED] wrote:
On May 5, 2007, at 12:06
On May 5, 2007, at 1:15 PM, Dave Miller wrote:
Adam Jacob Muller wrote on 5/5/07 1:06 PM:
Hi,
I have some annoying telemarketer calling me on a recurring basis,
but
I'd like to discourage them a bit and have some fun.
I found this:
Tzafrir Cohen wrote:
On Sat, May 05, 2007 at 06:23:43PM +0200, Remco Post wrote:
Mark Coccimiglio wrote:
Tzafrir;
Actually I have found this config to work really well. I prefer to
use a script run from inittab but Ubuntu doesn't work like Redhat or
BSD. On a production box keeping
SLA requires meetme which requires at a minimum ztdummy. So, you must
compile and install zaptel, then compile and install asterisk 1.4.3 and
the sla commands will be in the CLI. Let me know if you need help
setting up SLA on Polycom phones with *. I've done it successfully and
have the
Hello All,
I just received some test units of Grandstream GXV-3000 IP Video Phone.
I did some research and looks like Asterisk 1.2 does not support video
H.264 but Asterisk 1.4 does. Is it correct?
Actually I did try to test with Asterisk 1.2 and video did not
initialize but voice worked...
Hi Salvatore, thanks for reply.
And if pabx legacy was Siemens model HiPath 3750, could use asterisk as
serving of voicemail and other applications?
Best Regards
Josué
2007/5/5, Salvatore Giudice [EMAIL PROTECTED]:
I've done a lot of work with Avaya. Voicemail systems attaché dot Avaya
use
Thanks I did that as well.
I did however get the problem fixed by setting canreinvote=yes
Apparently the polycom wants it when the soft phones don't
Sorry, I meant canreinvite can re inVOTE- is something that dead people do
in my home state of Mississippi. G
-Original Message-
From:
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Este mensaje ha sido analizado con NOD32 antivirus system
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Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de
mexico, asi que en parte tienes razón, pero tb creo que deberías haber
puesto de donde eres.
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Rodrigo
Mercado
Enviado el: sábado, 05 de mayo de 2007 12:38
Para:
Ed Nuñez wrote:
Does anyone know how to gain access directly to the configuration files
in AsteriskNow? I have dual NICs and need to change the binding in the
config file. I believe they blocked ssh2 access by default.
ssh is not blocked. You have to ssh into the userid admin. If
On Sat, May 05, 2007 at 02:25:50PM -0500, Cesar Benjamin Garcia Martinez wrote:
My question is becouse i read than 1.4 supports T.38, and then should
receive fax i guess...
1.4 only supports VoIP passthrough of T.38 . That is: if you get a T.38
fax, it can be safely redirected to a
At the very least, he's abusing his customers. Substances? I hadn't
thought of that.
On 4/30/07, Salvatore Giudice [EMAIL PROTECTED]
wrote:
I suspect that Jed has a substance abuse problem and that he may be in
rehab. I don't know for sure of course. This kind of silence is indicative
of
Dear All,
Why does my configure fail like so:
checking for pg_config... /usr/local/pgsql/8.2.4/bin/pg_config
checking for PQexec in -lpq... no
configure: ***
configure: *** The PostgreSQL installation on this system appears to be broken.
configure: *** Either correct the installation, or run
On Sat, 5 May 2007, Bill Merriam wrote:
To get root access type sudo su. Once there you can change the root
password with passwd.
Seems a bit redundant:
-fs::sedwards:~$ man sudo
NAME
sudo - execute a command as another user
-fs::sedwards:~$ man su
HI
I have 3 Linksys SIP901 IP phones
I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd
I'm looking to connect this phones together and to make calls between them
Not from outside of my lan
I don't know how to configure asterisknow beta
Can somebody help
I'm
Adam Jacob Muller wrote on 5/5/07 1:38 PM:
On May 5, 2007, at 1:15 PM, Dave Miller wrote:
Adam Jacob Muller wrote on 5/5/07 1:06 PM:
Hi,
I have some annoying telemarketer calling me on a recurring basis, but
I'd like to discourage them a bit and have some fun.
I found this:
Ardit Saliu wrote:
HI
I have 3 Linksys SIP901 IP phones
I also have a pc I’m not using it amd athlon 1800+ 512mb ram and 40 gb hdd
I’m looking to connect this phones together and to make calls between them
Not from outside of my lan
I don’t know how to configure asterisknow
HI
I have 3 Linksys SIP901 IP phones
I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd
I'm looking to connect this phones together and to make calls between them
Not from outside of my lan
I don't know how to configure asterisknow beta
Can somebody help
I'm
Dear All,
Where can I find a res_pgsql.conf and some sql to insert for tables etc.?
Are all db res things to be done via odbc now?
Why was this included with no docs or sample conf?
Thanks,
Gavin.
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Hey
Look
http://www.asterisk-es.org
Best Regards
On 5/5/07, Cesar Benjamin Garcia Martinez [EMAIL PROTECTED] wrote:
Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de
mexico, asi que en parte tienes razón, pero tb creo que deberías haber
puesto de donde eres.
No matter what I do, ${ANSWEREDTIME} is always 0, even on the most
simplest dial plan such as:
Using Asterisk 1.2.13
exten = 77,1,Answer
exten = 77,2,Playback(custom/dax/S300) ; one minute file
exten = 77,3,Noop(${ANSWEREDTIME})
exten = 77,4,Hangup
-- Executing Answer(SIP/5402-b7b45f58, )
To everybody: Thanks for your thoughts and suggestions. This will be my
last post to this list on this subject.
I've started a blog about my research into this project:
http://myossjourneys.blogspot.com/
If you want to discuss this any further please do so over there.
Thanks again!
-MC
Hi,
I am in the process of planning a dial plan, In regards to the
requirement, I am confused how to go about the dial plan.
The scenario is like below.
BRANCH - A - (COMPANY)
Line 1 -- Extension 239
Line 2 -- Extension 8239
BRANCH - B - (COMPANY)
Hello guys, I am using the API Mananger with PHP to initiate a call from a
webpage. First I call to a line number, and then to an asterisk extension.
I followed examples on using the API Mananger, without any problem, and working
great.
Now I have a problem. I can initiate the call. I can call
steve,
thats Great... my C is old and ftw operated differently on sysV,
solaris, sunos, ultrix, and osf... so I went back to bourne...
couldn't work through the idiosyncracies of gnu autoconf, etc...
although I have a many reasons to, I just couldn't get to production 'C'
coding level...
options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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has anyone run into this message? for some reason, which I can not
determine, this script stop working and now gives this error. I googled
'outgoingspoolfailed' but not too much turned up... only questions, no
answers... :(
I am mv'ng a .call file to the ./outgoing directory. the call
nitesh,
you are correct. you need 1.4.x...
daveC
Nitesh Divecha wrote:
Hello All,
I just received some test units of Grandstream GXV-3000 IP Video Phone.
I did some research and looks like Asterisk 1.2 does not support video
H.264 but Asterisk 1.4 does. Is it correct?
Actually I did try
shawn,
you can set an archive variable in the .call file to 'yes' and it will
save it in ./outgoing_done... if there is now outbound line availible,
the .call file is updated (appended to) as per the status... * will keep
trying till it completes the calls or the number of retries is reached.
Has 1.2.19 been released ?
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: undisclosed-recipients:
Sent: Friday, May 04, 2007 12:20 PM
Subject: [asterisk-users] ASA-2007-013: IAX2 users can cause
unauthorizeddata disclosure
Asterisk Project
Its basically the same problem. Asterisk is not a standalone voicemail
server. It would have to support Qsig. Asterisk doea not exactly have
expansive Qsig support. I believe there are several bounties out for Qsig.
Without Qsig, you would have to use parallel forking and ring the users
Just forward them to 1-800-big-dick or some other 800 toll free phone sex
line. They can't tell they've been forwarded. They'll figure it out
eventually.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
My money is on compulsory drug rehab or simply being held for 45 days of
observation after being caught sexually abusing a pony.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow
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