Asterisk compatible One to many video is achieved with VidPhone. You can
download the web embedded video component free by signing up an account on
my website www.itntelecom.com. Any help on usage, just send me a note and I
will be glad to help you set it up.
CS
-Original Message-
3 feb 2009 kl. 04.33 skrev Ex Vito:
On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno tipas...@gmail.com
wrote:
Is it possible to retrieve the DIALSTATUS variable when placing
call through
a call file. This variable is set when using the Dial()
application from
the dialplan, but I am
On Mon, 2009-02-02 at 22:25 -0200, Alejandro Cabrera wrote:
Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and
video. So I can establish a voip + video connection *one-to-one*
onlyit works OK.
But I'd like to implement a videoconference *one-to-many* in order to
Hi Lincoln,
Asterisk was expecting ACK after sending the 200 OK message. After
repeated attempts at sending the 200 OK message and not receiving ACK,
it terminated the call. Are you able to do a packet capture on the phone
end? Mostly likely the phone is sending the ACK, but its either sent to
bilal ghayyad bilmar...@yahoo.com writes:
My provider has one IP and one port ONLY, I need to send for him the
calls from different IP's on the same Asterisk machine, how?
You would have to run two instances of Asterisk to achieve that right
now.
1) You can't distinguish on source port
Hi,
I want to convert SIP call in H.323. My terminator have H.323 base and I am
using Asterisk, Please help me in this regards.
Regards,
Asif
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asterisk-users
I have problem with packet size of voip packets, in a big network.
what is the best monitoring tools and analyzer for this purpose?
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asterisk-users mailing list
To
Hi,
I keep getting the following on my slug no matter what version I try (1.2, 1.4,
1.6, 1.6svn):
...
make[2]: Leaving directory `/usr/portage/distfiles/svn/trunk/menuselect/mxml'
gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a
menuselect.o: file not recognized: File
My call file was calling an AGI application, and from with the AGI, I could
not get the DIALSTATUS, I will try to send it to the dialplan first, then
call my AGI from the dialplan and see what happen.
Thanks for your help
On Tue, Feb 3, 2009 at 3:35 AM, Johansson Olle E o...@edvina.net wrote:
On Tue, Feb 03, 2009 at 05:14:42AM -0800, Gunner wrote:
Hi,
I keep getting the following on my slug no matter what version I try (1.2,
1.4, 1.6, 1.6svn):
...
make[2]: Leaving directory `/usr/portage/distfiles/svn/trunk/menuselect/mxml'
gcc -o menuselect menuselect.o strcompat.o
Hi,
I keep getting the following on my slug no matter what version I try (1.2,
1.4, 1.6, 1.6svn):
...
make[2]: Leaving directory `/usr/portage/distfiles/svn/trunk/menuselect/mxml'
gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a
menuselect.o: file not
Gunner wrote:
Hi,
I keep getting the following on my slug no matter what version I
try (1.2, 1.4, 1.6, 1.6svn):
Do you build this on the NSLU2 machine?
Left-over files from an x86 build? Try 'make clean' .
This is native on the NSLU2. It was right from svn, but I just tried
a
Hello folks.
First of all, sorry for my English :-)
I want to build dahdi-2.0.0 rpm from source (i have to use this version,
because OpenVox A1200p driver works only with it).
I've made some changes in .spec file (added one patch and one source
section) and trying to build rpm:
rpmbuild
Have you tried gcc'ing the menuselect module separately? (or removing it;
this would let the bsh scripts recreate as needed?)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Drew Gibson
Sent: Tuesday, February
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a
Can anyone suggest a SIP phone that allows conferencing of more than 3
parties locally (i.e. not as part of a conference bridge)? I didn't
realize Polycoms are limited to three.
TIA,
j
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There are some good examples of this at voip-info.org. Shouldn't this be
handled by normal caller-id?
Anyhow, here's an AGI (PERL) example:
#!/usr/local/bin/perl
use Asterisk::AGI;
# the AGI object
my $agi = new Asterisk::AGI;
# send callback reference
my $rc = $agi-set_callerid('IM_A_CALLER');
Jeff LaCoursiere wrote:
Can anyone suggest a SIP phone that allows conferencing of more than 3
parties locally (i.e. not as part of a conference bridge)? I didn't
realize Polycoms are limited to three.
I think most hardphones are limited to three-party conferences; if they
support more, it
Why are you trying to do it locally and not on the asterisk server
itself?
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
-Original Message-
From:
Hi:
I cant figure it out why fake ring is heared when dialing through SIP
(Asterisk) ,it often it gives me fake ring but not always ,in some calls it
gives me real ring.
the dialplan is without 'r' option.
_
Hotmail® goes where
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or
--- On Sun, 2/1/09, Ex Vito ex.vitor...@gmail.com wrote:
I'm trying to do the same in the SPA8000 units but
without any luck. If anyone is doing something similar with
this device then I'd appreciate it if you could share
your relevant config options (dial pattern, etc.).
Not sure
On Tue, 3 Feb 2009, Jeff LaCoursiere wrote:
Can anyone suggest a SIP phone that allows conferencing of more than 3
parties locally (i.e. not as part of a conference bridge)? I didn't
realize Polycoms are limited to three.
Grandstream GXP 2020 claims to be able to do 5-way.
I know the GXP
Have a look at smartCID at www.generationd.com
Uses a simple mySQL database, allows for call blocking flag, reverse CID
lookup, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent:
The Asterisk development team has released dahdi-linux 2.1.0.4
This release is available for immediate download from
http://downloads.digium.com/pub/telephony/dahdi-linux.
This release fixes a regression from dahdi-linux 2.1.0 in which it was
possible for the kernel to panic when conferencing
Hi,
Anyone can tell me what this means?
[Feb 3 12:42:32] WARNING[12130]: chan_sip.c:3293 update_call_counter:
Inringing for peer 'test-peer' 0?
Regards,
Mike
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Replacing a Nortel system and the owner is simply used to doing the
conferencing himself from his phone. I've talked him into using a bridge
and transferring the callers to it. We made a non-published bridge for
this with no PIN required.
I would suggest the Grandstream, but they really
The per-port regulators would be non-isolated. Probably feeding off
an internal 48V bus.
Yes, so that will be 90 to 95% efficient, but it is fed from an isolated supply
that at best will be 90%, probably less. Those numbers must be multiplied,
giving 81 to 86% overall efficiency--and I am
Hi All,
I am doing some research on the intergration of Jabber and Asterisk.
I have tried Jabber Client Mode. It's cool and works fine.
But there's few information on the Component Mode.
What's the difference between these two mode?
I finished the configuration on jabber.conf and I am using
Thanks for the nudge in the right direction. Turns out to be a quirk in
distcc. I can now compile (sans distcc), though much slower. At least it is
working :)
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve J. Douglas
Sent: Tuesday, February 03, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No Reply
Hi, all. I'm getting a lot of
[Feb 3 13:56:36] WARNING[3721]
/usr/src/asterisk/spandsp/agx-ast-addons/app_rxfax.c: Channel T30 DONE
0.
in my log file, and incomplete fax reception. Any idea what might be
going on? I've googled a fair bit, but haven't seen anything leap out at
me.
Thanks,
Lincoln King-Cliby wrote:
-Original Message-
Then starting at packet 3217 there are a series 6 of ICMP
Destination unreachable (Port Unreachable) messages from the
Asterisk server to the phone, with an RTP packet from the Phone
to the Asterisk server before each Destination
Either your RXFAX or the other end (TXFAX??) is turning on debug or ECM.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, February 03, 2009 1:23 PM
To:
On 2/3/2009 17:34, Asterisk Team wrote:
The Asterisk development team has released dahdi-linux 2.1.0.4
This release is available for immediate download from
http://downloads.digium.com/pub/telephony/dahdi-linux.
This release fixes a regression from dahdi-linux 2.1.0 in which it was
possible
On Tue, Feb 03, 2009 at 06:50:26PM +0300, bee-beeep wrote:
/usr/src/redhat/RPMS/i386/dahdi-linux-kmdl-2.6.18-92.1.22.el5-2.0.0-56.RHL5.i386.rpm
As you can see, it builds me only kmdl rpm - i wonder this is kernel module,
like zaptel-modules. So, if I'm trying to install dahdi-tools rpm,
Hello List,
I have been working on a little PHP software that uses AMI's
UpdateConfig command in order to modify some of it's config files.
I was working with 'Asterisk 1.4.22.1' and everything was working.
After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type:
Hello All ,
On Tue, 3 Feb 2009, Shaun Ruffell wrote:
Thomas Kenyon wrote:
On 2/3/2009 17:34, Asterisk Team wrote:
The Asterisk development team has released dahdi-linux 2.1.0.4
This release is available for immediate download from
Remco Barendse wrote:
1.4.23.1 is quite badly broken and there are no significant new
features
There are no new features at all, actually. What problems are you having with
1.4.23.1? It doesn't accomplish much to say that it is quite badly broken
without at least telling what is wrong.
Mr. James W. Laferriere wrote:
On Tue, 3 Feb 2009, Shaun Ruffell wrote:
Thomas Kenyon wrote:
I can't get this to build, the following error is produced:
CC [M]
/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o
I am using outgoing call files. I typically see the ooh something
changed / timeout on a regular bases every second to be exact.
Then it stops until some other call event happens.
So I mv my call file to the outgoing spool directory, I am listening
to that message, another call file is mv'ed
Olivier wrote:
Hi,
voip-info.org http://voip-info.org is almost silent regarding
udptl.conf except with
Depending on your fax device (such as the Linksys 3102) you may have
to edit the udptl.conf file. The error correction type that is sent is
usually the culprit of many problems with
Tilghman Lesher schrieb:
filesystem
timestamps are only resolute to the second, not to any fraction of a second.
Depends on the file system.
http://en.wikipedia.org/wiki/Comparison_of_file_systems
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Shaun Ruffell wrote:
Thomas Kenyon wrote:
I can't get this to build, the following error is produced:
CC [M]
/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o
/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c: In
function 'xproto_get':
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10
manager.c:1760: warning: ignoring return value of âreadâ, declared with
attribute warn_unused_result
is this anything to worry about?
can i safely ignore it?
Thanks
Robb
Robert Boardman wrote:
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10
manager.c:1760: warning: ignoring return value of âreadâ, declared with
attribute warn_unused_result
is this anything to worry about?
can i safely ignore it?
Thanks
Robb
I may be
Thomas Kenyon wrote:
On 2/3/2009 17:34, Asterisk Team wrote:
The Asterisk development team has released dahdi-linux 2.1.0.4
This release is available for immediate download from
http://downloads.digium.com/pub/telephony/dahdi-linux.
This release fixes a regression from dahdi-linux 2.1.0 in
[Feb 3 12:42:32] WARNING[12130]: chan_sip.c:3293 update_call_counter:
Inringing for peer 'test-peer' 0?
A sip_peer object in Asterisk has an inUse and inRinging number associated
with
it to keep track of the number of lines in use and the number of lines
that
are ringing for a
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mark Wiater
Sent: Tuesday, February 03, 2009 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No Reply to
On 04/02/2009 00:24, Mark Michelson wrote:
Robert Boardman wrote:
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10
manager.c:1760: warning: ignoring return value of âreadâ, declared with
attribute warn_unused_result
is this anything to worry about?
can i
Shaun Ruffell wrote:
Thomas Kenyon wrote:
No, does it need to be for building them then?
It shouldn't, but the module_refcount function is only defined in
kernels that are configured to allow module unloading. This probably
needs a mantis issue to make sure the drivers build when the
Hi,
voip-info.org is almost silent regarding udptl.conf except with
Depending on your fax device (such as the Linksys 3102) you may have to
edit the udptl.conf file. The error correction type that is sent is usually
the culprit of many problems with ATAs and T.38 providers.
Can anyone elaborate
I'm not familiar with packets specific to Asterisk, but do have some
familiarity with general Ethernet traffic. The Host unreachable messages you
are getting is from the protocol stack in the Linux computer, and generally
means the traffic is being sent to a port that is not open--i.e. no
Mike wrote:
Hi,
Anyone can tell me what this means?
[Feb 3 12:42:32] WARNING[12130]: chan_sip.c:3293 update_call_counter:
Inringing for peer 'test-peer' 0?
Regards,
Mike
A sip_peer object in Asterisk has an inUse and inRinging number associated with
it to
Thomas Kenyon wrote:
Shaun Ruffell wrote:
Thomas Kenyon wrote:
I can't get this to build, the following error is produced:
CC [M]
/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o
/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c: In
function
1.4.23.1 is quite badly broken and there are no significant new
features
Better to revert back to 1.4.22.1
On Tue, 3 Feb 2009, Jose P. Espinal wrote:
Hello List,
I have been working on a little PHP software that uses AMI's
UpdateConfig command in order to modify some of it's config
On Tuesday 03 February 2009 16:26:51 Jerry Geis wrote:
I am using outgoing call files. I typically see the ooh something
changed / timeout on a regular bases every second to be exact.
Then it stops until some other call event happens.
So I mv my call file to the outgoing spool directory, I am
On Tuesday 03 February 2009 17:47:26 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
filesystem
timestamps are only resolute to the second, not to any fraction of a
second.
Depends on the file system.
http://en.wikipedia.org/wiki/Comparison_of_file_systems
Fair point, but what standard
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
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2009/2/4 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
gust...@utopixnetworks.com schrieb:
Me encuentro de vacaciones hasta el proximo 16/02.
Enjoy your vacation.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr.
Please get this out of office reply disabled, or at the very least, fixed...
It currently seems to have generated a loop, sending out of office replies
to the out of office replies it's already sent to the asterisk-users mailing
list... It's bad that it sent a reply to the list anyway, but this
Mark Michelson wrote:
Olivier wrote:
Hi,
voip-info.org http://voip-info.org is almost silent regarding
udptl.conf except with
Depending on your fax device (such as the Linksys 3102) you may have
to edit the udptl.conf file. The error correction type that is sent is
usually the
Tilghman Lesher schrieb:
On Tuesday 03 February 2009 17:47:26 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
filesystem
timestamps are only resolute to the second, not to any fraction of a
second.
Depends on the file system.
http://en.wikipedia.org/wiki/Comparison_of_file_systems
Not an Asterisk based solution but you can look at getting an Adtran Atlas
550 with PRI and BRI cards.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Daniel Harper
Sent: Sunday, February 01, 2009 11:02 PM
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
Me encuentro de vacaciones hasta el proximo 16/02.
Por favor contactar a Pablo Minsteras (pablo.minste...@utopixnetworks.com) o
Gustavo Hernandez (gustavo.hernan...@utopixnetworks.com).
Muchas Gracias,
Gustavo Scheveloff
___
-- Bandwidth and
gust...@utopixnetworks.com schrieb:
Me encuentro de vacaciones hasta el proximo 16/02.
plonk
Philipp Kempgen
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options
On Tuesday 03 February 2009 21:01:54 Philipp Kempgen wrote:
gust...@utopixnetworks.com schrieb:
Me encuentro de vacaciones hasta el proximo 16/02.
plonk
John Todd is at Digium|Asterisk World at ITEXPO in Miami Beach today, or
he would have noticed this and corrected it earlier. He has been
resea...@businesstz.com wrote:
Can someone assist me on this please?
Hello List
I am setting up a small demo site using SS7 and one of the requirement is
to be able to unhide the numbers and locate exact location of the caller
(BTS ID). Vodafone uses Nokia-Siemens switch and has
Hi Lincoln,
The fact that you can hear and respond to the voice mail (even if its
for the first 20 seconds), means that your phone has received the OK
message properly. The problem is the missing ACK after receiving OK.
When asterisk did not receive the ACK after a few retries of the OK, it
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'd like to know what is the current situation with regard to AOC-E,
when Asterisk is inserted between the telco and an existing PBX, using
E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the
telco to the PBX, so that billing
Ya lo sabemos, ya que hemos recibido unos 10+ mensajes de esto.
Por favor no envie mas correos electronicos asi y repare su programa de
notificaciones de vacaciones para que no envie correos duplicados.
gust...@utopixnetworks.com wrote:
Me encuentro de vacaciones hasta el proximo 16/02.
Thanks Matt
I will speak to voda to know exactly parameter name and let your know soon
Regards
Sam
resea...@businesstz.com wrote:
Can someone assist me on this please?
Hello List
I am setting up a small demo site using SS7 and one of the requirement
is
to be able to unhide the numbers
I would like to be able to stop the chanspy application and go to the next
step in the dialplan but I do not see a way to do that.
I have looked at the code and I do not see a way to stop the chanspy
application.
Even if there are no channels that match the chanprefix pattern the chanspy
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