Hello.
Is there a way to stop AGI Originate Action once it has started?
I'm trying to bridge incoming call to an extension, and if incoming call
hangs up
the phone on the extension keeps ringing until timeout. I've tried googling,
and
checking mailing list archive, but with no success.
I'm
I'd contact Digium - they're really good with providing support - just
add the following line and dial it:
Thanks Matt for your suggestion.
We despatched a new TE412P card to replace the existing card but the
same problem occurred. So, I think it is not the Digium card problem.
At the same
Ok.
Now, I follow the digium documents.
I have the TC400B-user-manual-1.pdf
Where have I to insert the mode=g729 ?
Chapter 3
Configuration
At this time no zaptel.conf or zapata.conf changes are necessary to utilize
this card. The mode module parameter may be used to
I have Asterisk 1.6.1.4 and GUI 2.0 (Latest). I never had such problem (my
main Asterisk server is at 1.6.0.6 with latest GUI as well).
I would reinstall the GUI. But I can tell you it *SHOULD* work.
Oh! While I'm writing this mail. I just looked at one of my client's
Asterisk: 1.6.1.1
Hi People
We have a client who want to route their outbound calls through our
asterisk server. We need them to authenticate as a sip extension so we
know which calls are coming through them but the people over their side
seem to be a bit clueless and claim they can't authenticate.
Does anyone
Hi,
I have a 4 port analog cards with asterisk 1.4.26.1 (centos5.3)
installed. After I dial an outgoing call, it returns error and call
drop as below. Anyone can tell me what the problem is. ango
-- Executing [8...@internal:20] Dial(SIP/601-09425ab8,
dahdi/g0/8200|50|T) in new stack
[Aug
On Wed, 19 Aug 2009, Terry Wilson wrote:
I haven't seen (or heard of) it happening. Please post a bug report
on http://betareports.digium.com/mantis/ with a backtrace from one of
the core dumps along with the relevant information about your setup,
dialplan, chan_skype.conf, etc. If there
No way to authenticate from Ser, it's a proxy...
Just make an entry in your asterisk sip.conf with the IP of SER and no
authentification.
Do the necesssary stuff in your extension.conf to identify and bill your
client.
Olivier
Ishfaq Malik a écrit :
Hi People
We have a client who want to
The authentication is in the SER and asterisk trust the ser (insecure=invite)
Just do some iptables to be sure to don't receive sip from others...
Cordialement,
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De :
As subject really - anyone using a Patton ISDN2e box purely as a SIP to
ISDN2e bridge, rather than use it's router, etc. functions?
Just looking at options for an existing client who has a VoIP only box
who's moved to a location with somewhat challenging broadband
connectivity...
(This is in
Hi
I know this is far from best practice but is it possible to authenticate
a sip peer on the IP address it's coming from so that it doesn't need to
use a UN and Pass?
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
___
BERGANZ François wrote:
Where have I to insert the mode=g729 ?
That depends on the platform. On Ubuntu I added it to
/etc/modprobe.d/dahdi:
options wctc4xxp mode=g729
--
Sean Bright
sean.bri...@gmail.com
___
-- Bandwidth and Colocation
Ishfaq Malik schrieb:
Hi
I know this is far from best practice but is it possible to authenticate
a sip peer on the IP address it's coming from so that it doesn't need to
use a UN and Pass?
Yes. That's exactly what type=peer is for.
regards
klaus
In the sip.conf, just insert the host=
Cordialement,
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Ishfaq Malik
Envoyé : mardi 25 août
Hi
I've created a peer that only uses IP Address for authentication but now
I'm being told that SER can't generate a register command, is this true?
Ish
hh174 wrote:
No way to authenticate from Ser, it's a proxy...
Just make an entry in your asterisk sip.conf with the IP of SER and no
Yes, it's true, SER is SIP proxy. It can't generate anything. It
only forwards requests and replies.
Ishfaq Malik wrote:
Hi
I've created a peer that only uses IP Address for authentication but now
I'm being told that SER can't generate a register command, is this true?
Ish
hh174
Try us...@lists.kamailio.org
http://www.acropolistelecom.net
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Ishfaq Malik
Envoyé : mardi
SER can't generate REGISTER
You can play to control if it is yours with adding headers if you want, we can
imagine that...
Cordialement,
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
Hello Asterisk users around the world!
Recently, I have been working with pretty large Asterisk
installations. 300 servers running Asterisk and Kamailio (OpenSER).
Replacing large Nortel systems with just a few tiny boxes and other
interesting solutions. Testing has been a large part of
Hello,
I have a problem of DTMF duplication.
I receive call from my provider with SIP protocol. These calls pass
through an interactive voice menu, using the application Waitexten to
enter a client code. The menu works fine, but sometimes I have DTMF
duplication that prevent proper code
IPKall does not alter the CLID number, what it recieves, it forwards on.
Contact your carrier to find out why it was delivered incorrectly.
IPKall does not provide Voice Mail, and will not answer the call if the VoIP
destination un-reachable.
IPKall
_
From:
Hi,
what machines where the IBM Servers? I would be really interested in
this as we have currently IBM Hardware deployed and, well, maybe it's
time to investigate in different hardware,
best
--
Raimund Sacherer
-
RunSolutions
Open Source It Consulting
-
Email: r...@runsolutions.com
Hello,
I am developing an asterisk autodialer. I am looking for the following
information:
1. Detailed Configuration Documentation for Asterisk Autodialer
2. Volume Testing Strategy
3. Lessons Learnt from past Asterisk Autodialer configuration
4. What are the different asterisk
On Tue, Aug 25, 2009 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote:
Hello Asterisk users around the world!
Recently, I have been working with pretty large Asterisk
installations. 300 servers running Asterisk and Kamailio (OpenSER).
Replacing large Nortel systems with just a few tiny
I would be curious to know if bonding 2 Ethernet ports together would help
to push the upper limit a bit further ...
(by the way, this limit is 11000 channels or 5500 calls, isn't it ?
___
-- Bandwidth and Colocation Provided by
On Tue, Aug 25, 2009 at 10:04 AM, Sanjoy Rath sanjoy_r...@hotmail.comwrote:
Hello,
I am developing an asterisk autodialer. I am looking for the following
information:
1. Detailed Configuration Documentation for Asterisk Autodialer
2. Volume Testing Strategy
3. Lessons Learnt from past
Hi
Someone may give me an example of followme() application using in a dialplan
(including what to configure in followme.conf) ?
I use asterisk 1.6.1 so if your example can match to that release it's will
be wonderfull.
Thank in advance.
Harry.
___
--
www.IPKall.com wrote:
IPKall does not alter the CLID number, what it recieves, it forwards
on. Contact your carrier to find out why it was delivered incorrectly.
That has NEVER been the way IPKall worked
IPKall arrived, until a week or more ago, with my IPKall number, then it
For someone who is developing an 'autodialer' you are asking for an awful
lot! I would recommend getting to grips with asterisk before even
considering developing a dialer...
question 1 - aren't you developing your own so why would you need
documentation for another? or... why not use the other?
That was quite a helpful reply.
Cheers,
SR.
Date: Tue, 25 Aug 2009 15:40:20 +0100
From: gera...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Autodialer
For someone who is developing an 'autodialer' you are asking for an awful lot!
I would
Thanks Steve for helpful reply.
Cheers,
SR.
Date: Tue, 25 Aug 2009 10:26:37 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Autodialer
On Tue, Aug 25, 2009 at 10:04 AM, Sanjoy Rath sanjoy_r...@hotmail.com wrote:
If you don't see why you are getting replies of this colour, you must
have an ingrained inability to understand what does and does not
constitute a question of a manageable scope. It must be sufficiently
narrow and specific in order to be realistically addressable in this
forum.
You can't
Hello everybody.
I want know. witch version of linux i can used from install asterisk..
Grettings..
___
Jorge Rodriguez
Coordinador
Club Amigos De Torreon
www.amigosdetorreon.com.mx
Cel: 8711684631
Mr. Rodriguez wrote:
I want know. witch version of linux i can used from install asterisk..
There is only one Linux. You must be referring to Linux distributions,
which are a way of packaging Linux with other software, utilities,
package management, administrative tools, etc.
In that case,
Alex,
You are right. My questions are probably wide open. I probably would have more
specific. Any help I get would be great; if not I will figure it out myself :).
Steve, Geraint - It was kind of you both to respond back. Thanks guys for your
response. :)
I am dCAP trained but
Mr. Rodriguez wrote:
I want know. witch version of linux i can used from install asterisk..
On Tue, 25 Aug 2009, Alex Balashov wrote:
There is only one Linux. You must be referring to Linux distributions,
which are a way of packaging Linux with other software, utilities,
package
Here is my .02; There are basically (but not only) two ways to auto-dial
from asterisk. You can use Asterisk Manager or create a set of call files
to use the pbx_spool function. I'm more familiar with the latter. If you
dump a set of call files, Asterisk will try to do them all at once and
On Tue, 2009-08-25 at 16:28 +0200, harry R wrote:
Hi
Someone may give me an example of followme() application using in a
dialplan (including what to configure in followme.conf) ?
I use asterisk 1.6.1 so if your example can match to that release it's
will be wonderfull.
snip
We are using
Hi Danny,
Thanks so much for your response. I have used call files to autodial too. Can
this be used for large volume calls?
Thanks,Sanjoy.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 25 Aug 2009 11:07:04 -0500
Subject: Re: [asterisk-users] Asterisk Autodialer
Sanjoy Rath escribió:
Anyways I checked VOIP-info.org the information there was pretty
basic. I was trying to get some more insight to this autodialer stuff.
If there is something I can take leverage of that will be great
(because I do not want reinvent the wheel) or else (as sadi before) I
On Tue, 2009-08-25 at 12:21 -0400, John A. Sullivan III wrote:
On Tue, 2009-08-25 at 16:28 +0200, harry R wrote:
Hi
Someone may give me an example of followme() application using in a
dialplan (including what to configure in followme.conf) ?
I use asterisk 1.6.1 so if your example can
25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further ...
(by the way, this limit is 11000 channels or 5500 calls, isn't it ?
Yes, this is 11.000 channels.
Bonding is good advice, provided we
That would be a qualified yes. The problem you will face is this; Say you
want to do 1000 calls and you have 25 lines you can use. If you dump all
1000 calls into /var/spool/asterisk/outgoing at once, asterisk will try to
process all 1000 at once, which will lead to some problems. The way to
AMI is definitely the more sophisticated and less naive way to
overcome these problems, and will definitely provide the maximum level
of event feedback and call control possible. Call files are a much
more fire and forget strategy, and getting pacing and overdial ratio
right is a lot
On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further ...
(by the way, this limit is 11000 channels or 5500 calls, isn't it ?
Danny,
Thanks for your response.
Thanks,Sanjoy.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 25 Aug 2009 11:34:29 -0500
Subject: Re: [asterisk-users] Asterisk Autodialer
That would be a qualified yes. The
problem you will face is this; Say you
Thanks Miguel. Have your configured GNUDialer before?
Date: Tue, 25 Aug 2009 11:22:16 -0500
From: mmol...@millenium.com.co
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Autodialer
Sanjoy Rath escribió:
Anyways I checked VOIP-info.org the information there
25 aug 2009 kl. 18.50 skrev John A. Sullivan III:
On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further ...
(by the way, this limit is
As said before, SER is a proxy, it will not be able to register or
authenticate.
Just do your own stuff to be sure it's your client.
Olivier
Ishfaq Malik a crit:
Hi
I've created a peer that only uses IP Address for authentication but now
I'm being told that SER can't generate a
I would prefer to use AMI. Let me start looking into AMI. I would like to
include functionalities like upload numbers to call from an interface, i want
reports back numbers called, setup call time etc. Let me look up AMI. Thanks
Alex for the info.
From: abalas...@evaristesys.com
To:
Sanjoy Rath wrote:
I would prefer to use AMI. Let me start looking into AMI. I would like
to include functionalities like upload numbers to call from an
interface, i want reports back numbers called, setup call time etc. Let
me look up AMI. Thanks Alex for the info.
Yep. If you need that
On Tue, Aug 25, 2009 at 12:28 PM, Olle E. Johansson o...@edvina.net wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further ...
(by the way, this limit is 11000 channels or 5500 calls,
Searching their support forum, posted today is the fact they are
discontinuing any VM
The message saying that they are discontinuing their offering of
voicemail was posted on August 24, 2007 - two years ago. That
doesn't seem to be a new issue.
On Tue, Aug 25, 2009 at 1:29 PM, Alex Balashov abalas...@evaristesys.comwrote:
Sanjoy Rath wrote:
I would prefer to use AMI. Let me start looking into AMI. I would like
to include functionalities like upload numbers to call from an
interface, i want reports back numbers called, setup call
Pascal Bruno wrote:
I do not quite agree, I have developed a system exactly like that using
call files, and I do have an interface to upload the numbers to call,
you can setup call time, and the the delay to wait between each call.
To me it was very straight forward and it works great,
On Aug 24, 2009, at 10:49 PM, Michel Verbraak wrote:
snippage of stuffs
As I see you have specified nl as defaultzone so I expect that you
are using a ISDN-30/15 line from provider KPN in the Netherlands.
If so then remove the crc4 option from the span line in /etc/dahdi/
system.conf.
On Tue, Aug 25, 2009 at 2:52 PM, Alex Balashov abalas...@evaristesys.comwrote:
With enough spiritual commitment, anything can be done; you certainly
*can* do it this way. You can write a fairly sophisticated dialer in
Bash, too.
The issue is whether it is methodologically correct and
25 aug 2009 kl. 19.42 skrev Steve Totaro:
On Tue, Aug 25, 2009 at 12:28 PM, Olle E. Johansson o...@edvina.net
wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
I would be curious to know if bonding 2 Ethernet ports together
would help to push the upper limit a bit further ...
(by the
Hello there!
I was testing Asterisk for the last two weeks using the Realtime driver
for MySQL, and leaving rtcachefriends=yes configured to enable MWI.
Today I started making additional tests with rtcachefriends=no because
we will probably need to use Asterisk without this cache.
For some
On Tue, Aug 25, 2009 at 3:15 PM, Pascal Bruno tipas...@gmail.com wrote:
On Tue, Aug 25, 2009 at 2:52 PM, Alex Balashov
abalas...@evaristesys.comwrote:
With enough spiritual commitment, anything can be done; you certainly
*can* do it this way. You can write a fairly sophisticated dialer
Steve Totaro wrote:
Agreed, simple cron jobs can count and distribute call files across
servers with very little scripting.
It is simple, and simple and efficient, and that is good.
If the requirements are also simple, I'd be the last to argue with the
simplest approach.
In every case in
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.
I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
TE121B PCI express card with a HARDWARE echo cancellation module. All
this is housed on a CentOS 5.5 box, 2.6.18 Kernel. My incoming phone
On Tue, 25 Aug 2009, Bharath B. Reddy Bynagari wrote:
I am pretty new to Asterisk. I am trying to make sure some human being
answers the phone not the voice mail machine. How can I programmatically
identify that?
The best way is to ask the caller to press a key to continue. Most
answering
On Wed, 2009-08-26 at 06:18 +1000, Alex Samad wrote:
On Tue, Aug 25, 2009 at 07:30:08PM +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 18.50 skrev John A. Sullivan III:
On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
[snip]
On Tue, Aug 25, 2009 at 07:30:08PM +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 18.50 skrev John A. Sullivan III:
On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote:
25 aug 2009 kl. 16.20 skrev Olivier:
[snip]
mode
in Linux on any old switch and it works reasonably well
Hi,
I am pretty new to Asterisk. I am trying to make sure some human being
answers the phone not the voice mail machine. How can I programmatically
identify that?
Here is my Sub:
sub DialPhysician {
my ($self, $con, $PhysicianPhone, $call_id, $conv_id) = (@_);
Jason,Echo Hellooo old buddy! Do you evefr go 4 the dCap?
Your bud from the MD class.
Al
On Tue, Aug 25, 2009 at 4:43 PM, Jason Baker jba...@glastender.com wrote:
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.
I am running Asterisk 1.4.25 with Dahdi
Hello there!
Problem found.
For some reason, the update statement below is generated with an invalid
atribution of empty value '' to field port that is an integer.
Because of that, this record keeps with prior fullcontact information
that was updated by another client (which uses a different
Jason Baker wrote:
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.
I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
TE121B PCI express card with a HARDWARE echo cancellation module. All
this is housed on a CentOS 5.5 box, 2.6.18 Kernel. My
I did with Gnudialer.
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
From:
Hello, all. Since implementing an iptables firewall between the
Asterisk PBX and several SIP phones, the Asterisk PBX ability to
reinvite has been broken even when the phones are on the same network
(i.e., no firewall between the phones). We've been beating our heads
against the wall thinking it
On Tue, Aug 25, 2009 at 12:50 PM, John A. Sullivan
IIIjsulli...@opensourcedevel.com wrote:
You don't necessarily need a switch to support it. One can use alb mode
in Linux on any old switch and it works reasonably well other than for
some excessive ARP traffic. However, as we found out the
On Tue, 2009-08-25 at 21:57 -0400, David Backeberg wrote:
On Tue, Aug 25, 2009 at 12:50 PM, John A. Sullivan
IIIjsulli...@opensourcedevel.com wrote:
You don't necessarily need a switch to support it. One can use alb mode
in Linux on any old switch and it works reasonably well other than for
Folks,
Had a request from a customer: is it possible for a customer, using a
password to restrict others from making long distance/cell calls?
That is, the user set a level of service?
Something like this:
Customer dials a number -- operator asks for password, then service
level (another
Hi All,
suppose this:
Dial(SIP/somecarrier/somenumber/60/L(360)M(td|${EPOCH})
where 60 is the seconds to wait for the callee (the called party) to answer
L(360) is the absolute limit of the call once it has been answered, in ms
M(td|${EPOCH}) is the macro to execute when the call gets
Hellow,
i need the following requirements with asterisk :
1) Can ACD (Automatic Call Distribution) service work with asterisk, and
how to set up ACD in asterisk ?
2) How call barging can set up in asterisk ?
3) How call recording can set up in asterisk?
Thanks
mahboob
System Engr
SSL
1) Can ACD (Automatic Call Distribution) service work with asterisk, and how
to set up ACD in asterisk ?
You can (and it is better to) write your own code in Asterisk.
2) How call barging can set up in asterisk ?
There is a zap barge cmd - not sure if this is what you want.
3) How call
77 matches
Mail list logo
