Dear all
I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem
i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make
outgoing call it is working fine but incomming call not handling ...when i call
from outside to this line it is rinning but
Hi all, I have a question on asterisk conference.
Now I use appl Meetme with option A x because when a marked person
hangup I want to close all the conference.
But what I have to do if I want two marked person and kill the
conference when one of two hangup?
Is possible?
Thanks. Enrico.
--
Atis Lezdins wrote:
On 1/18/08, Cyril SCETBON [EMAIL PROTECTED] wrote:
Hi guys,
Does someone use a mysql database for accessing data and in the same
time for storing cdr ? if that is the case, which module is used ?
There are two different modules for this. But it's all in
Andrea Spadaccini wrote:
Ciao Cyril,
Does someone use a mysql database for accessing data and in the same
time for storing cdr ? if that is the case, which module is used ?
thanks
Which kind of data are you talking about?
I suppose that you mean that you want to store non-Asterisk
Hello,
I use Asterisk version 1.2.7.1.
A user who uses my Asterisk made me part of a worry about listening to his
voicemails. He has received 4 voicemails on January 3, respectively at 3.00
pm, 3.36 pm, 3.41 pm and 4.40 pm. He has received notifications by e-mail at
these times.
On first
In my series of articles about Asterisk 1.4, I've now arrived to the
new jitter buffer
that enhances voice quality for those of you using Asterisk as a PSTN
gateway.
Please read
http://www.voip-forum.com/category/asterisk/asterisk14/
/O
___
--
Michael J. Liberatore [EMAIL PROTECTED] writes:
I do have queues set up but I would have to setup queues for all calls
then, even from other inside the office calls. Cause if I disable call
waiting, wouldn't that be the same as saying maximum sip connections to
the phone = 1?
Call waiting
Hi to all,
i already searched the archive without finding a solution to my problem.
I have asterisk installation 1.2.18 to support multiple virtiual PBXs.
I use SIP peer in the format ID-EXT to let every virtual PBX to
share the same numbers of EXT.
Ex.
(PBX ID 10 Extensions)
10-101
10-102
bilal ghayyad a écrit :
Hi All;
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot, but we need also the
ability to use IAX (as it is good for NAT).
We are using IP0027. Great audio, POE, 5 SIP accounts, 1
On 1/22/08, Michael J. Liberatore [EMAIL PROTECTED] wrote:
I do have queues set up but I would have to setup queues for all calls
then, even from other inside the office calls. Cause if I disable call
waiting, wouldn't that be the same as saying maximum sip connections to
the phone = 1?
Or
If your working with Virtual PBX then why not set your users with there own
rules and normal extension numbers in there own context. You can have many
context.
That way only extensions you allow to see the context there in will have
those options.
- Original Message -
From: Marcello
Hi,
it is already that way but my config is made this way because i need
unique SIP peers to dial at.
May be i'm missing something but as i know is not possible to restrict
SIP peers (in sip.conf) in different context.
It should me made under multidomain support (one domain for each
peer).
Based on some rapid checks, 7.1.30 firmware behaves in exactly the same way.
Cheers,
Steve
On 1/22/08, Michael J. Liberatore [EMAIL PROTECTED] wrote:
Wow thanks so much for this, this is a lot of great info. Hopefully
enough to catch snom's attention to. Is it possible for you to try 7.x
on
Hello,
any one advise a good, strong and free softphone that can work with SIP
or/and IAX lines and supports attended transfer ?
Thanks for help.
Mit freundlichen Grüßen / best regards
André Herrlich
IT-Operator / Developer
LetMeRepair
LMR Service and
Hi all,
I am not getting the dial tone when i dial the zero digit.
And i am using analog card,for my operator phone caller id is not displaying on
the phone.I am in india.
In india is it possible to get the caller id for analog cards.
Can any body help me.
Please reply.
ThanksRegards,
Hi Cameron,
I think this paragraph goes to the heart of the matter...
Cameron Hissey wrote:
are connected to the standard switch, however the cabling was a bit
of a rush job and consequently the PoE has proven unstable on many of
the points, with some of them not even supplying data packets.
On 1/22/08, Steve Johnson [EMAIL PROTECTED] wrote:
Hi list,
There are many Polycom experts on this list -- hopefully someone has a
solution.
With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk
causes the Polycom 601 phones to start dumping these messages to the
CLI.
On Sun, 2008-01-20 at 00:57 -0800, bilal ghayyad wrote:
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot, but we need also the
ability to use IAX (as it is good for NAT).
I received a couple of ALL7960 phones
Going on 2 days now, without incident. 1.2.26 is by far the best update
I've done. Usually I end up rolling back within a few hours because of
show-stopping bugs.
On Jan 21, 2008 3:11 PM, Matt [EMAIL PROTECTED] wrote:
We have now been running 1.2.26 for the better part of today, without
I am using Polycom's SIP 2.2.0047 (the current release) and am seeing
this. It seems to occur less often with extensions reload rather
than just reload, but it would be nice to fix this.
Tx.
On Jan 22, 2008 8:30 AM, Steve Davies [EMAIL PROTECTED] wrote:
On 1/22/08, Steve Johnson [EMAIL
I've been reading up on followme app for asterisk 1.4 and I have it
working but I was wondering if the following was possible:
Based on followme.conf present the caller with the option to locate the
person:
Call comes in (external or internal) and rings extension with followme
configured.
In the dialplan you would just add a prompt and ask the caller to
press 1 to locate or hold for voice mail. If they press 1 launch the
followme app.
On Jan 22, 2008 10:25 AM, Anciso, Roy [EMAIL PROTECTED] wrote:
I've been reading up on followme app for asterisk 1.4 and I have it working
but
I've thought about that but is there a way to do this on whether or not
they are configured in follow.conf. I didn't want to introduce
unnecessary prompts for callers trying to reach people without followme
enabled.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Andre Herrlich wrote:
any one advise a good, strong and free softphone that can work with SIP
or/and IAX lines and supports attended transfer ?
IMHO there are no good softphones - at least not for
Mac OS X and I think that is true for Linux as well.
They're either not stable or their
Hello,
Have you tried our Zoiper softphone yet (www.zoiper.com) - new version
scheduled for in a couple of days ? If so, can you send me any remarks
of list so that we can keep those things in mind for future versions ?
Greetings,
Joachim
Philipp Kempgen wrote:
Andre Herrlich wrote:
what is the difference between FreeSwitch and Asterisk , whitch one is more
scalable and reliable?
_
Express yourself instantly with MSN Messenger! Download today it's FREE!
On Tue, Jan 22, 2008 at 04:04:29PM -0200, equis software wrote:
Hi!
Is there any way to login an agent by console command?
I want to login an agent doing this system call.
asterisk -rx 'AgentCallbackLogin 304 [EMAIL PROTECTED]'
Any ideas, thanks.
Something of the sort of:
originate
I have a little problem with mi outside calls. I Have Echo. It`s randomly.
I install codec G729 and I have Grandstream GXP2020.
Asterisk 1.4.9,Unicall 0.0.5pre1,Zaptel 1.4.5
I have a E1 and 30 channel.
Can you give a tip, where can I check If I miss something.
Hi!
Is there any way to login an agent by console command?
I want to login an agent doing this system call.
asterisk -rx 'AgentCallbackLogin 304 [EMAIL PROTECTED]'
Any ideas, thanks.
___
-- Bandwidth and Colocation Provided by
Hello everybody,
I'm running Asterisk 1.2.24 on three servers which are configured
almost identical. The servers use IAX to communicate between each other
and SIP to communicate with the outside world through a Patton
Smartnode 4960 gateway. One server has about 30 SIP phones registered,
the
Please read:
http://www.voip-info.org/wiki/view/FreeSwitch
and
http://www.voip-info.org/wiki/index.php?page=Asterisk
Then if you have a specific question about one of them,
come back here to ask about asterisk, and on the freeswitch
mailinglist for more info on that technology.
Or you could
I have a client that is ready to make the step to a local Asterix
unit, but I don't have the bandwidth to take this project on by
myself.
(They currently have 2-3 POTS lines which they wil be using as trunks).
--
- Brian Gupta
___
-- Bandwidth and
On Tuesday 22 January 2008 11:50:25 love U.all wrote:
what is the difference between FreeSwitch and Asterisk ,
The main difference in functionality is that FreeSwitch is a voip-switch only.
It does not have any method to interface to the PSTN, other than through
using another host which does
what is the difference between FreeSwitch and Asterisk ,
The main difference in functionality is that FreeSwitch is a
voip-switch
only.
Technically, FreeSWITCH is a soft-switch, or a modular media switching
library that can switch more than just voice. Also, technically, FS is
a library,
Hi everyone,
I have a few asterisk machines doing PSTN calls, and I keep track of
all cdr in a single machine running mysql 5. Since I have a very
large amount of records in there, its getting pretty slow to query
the database, so I'm wondering if anyone does some type of log
rotating, like save
On 1/22/08, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi everyone,
I have a few asterisk machines doing PSTN calls, and I keep track of
all cdr in a single machine running mysql 5. Since I have a very
large amount of records in there, its getting pretty slow to query
the database, so I'm
I have just retested and agree that this error eventually does clear
itself. However, in this test it took about 35 minutes and each
Polycom phone produced between 1000 and 1300 error message lines at 1
to 0 second intervals (which I captured to the debug log). Once one
phone starts flagging an
Dear Jared;
Thanks a lot, what is the website link (from where I
can buy this device)? Also, do u have any idea about
the prices?
Regards
Bilal
--
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot,
Hi,
Is it possible to dial voicemail from a particular phone line and
automatically enter the extension that is being dialed from, thereby only
prompting for the password?
I've been searching around to find if this is possible, but I haven't been
able to find an example of this. I have a feeling
My guess is you want the server to call the user and play the voicemail?
On Jan 14, 2008 1:37 PM, Gilberto Nunes [EMAIL PROTECTED] wrote:
A Monday 14 January 2008 16:25:15, Steve Johnson escreveu:
Yeah! I'm just do this right now!
But I want more!
How can I create some extension to call
Heres what I do for this:
exten = *85,1,VoicemailMain(${CALLERID(NUM)})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of arkda
Sent: Tuesday, January 22, 2008 5:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
VoicemailMain(${CALLERID(number)}) is probably what you want.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons
___
-- Bandwidth
I once attended an office with such bad cabling that we put the switch
on top of the server and ran cables against the walls to prove that the
internal cabling was rotten.
PaulH
On Tue, 2008-01-22 at 12:26 +1100, Cameron Hissey wrote:
After changing all the networking and removing PoE, the
You may speed up your queries with proper indexing. The default indexes are
included with the table creation script here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
ALTER TABLE `cdr` ADD INDEX ( `calldate` );
ALTER TABLE `cdr` ADD INDEX ( `dst` );
ALTER TABLE `cdr` ADD INDEX (
You would have to write an external app to create a call file after each
vm is left...probably doable (externap in voicemail.conf), probably
fiddly.
PaulH
On Tue, 2008-01-22 at 17:44 -0500, arkda wrote:
My guess is you want the server to call the user and play the
voicemail?
On Jan 14,
Unreachables are the the sign you are looking for!
We had a client where that happened all the time and after we
disconnected the phone network from the pc network it all went away.
Never bothered looking into it further.
PaulH
On Wed, 2008-01-23 at 10:49 +1100, Cameron Hissey wrote:
Well
Hah, thanks guys. I had tried that previously, but my syntax was off.
On Jan 22, 2008 6:19 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
VoicemailMain(${CALLERID(number)}) is probably what you want.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details
For anyone who cares to know. I finally got it working correctly. Turned
out I needed fromuser set. Then it was just playing around until it
started working.
register = [EMAIL PROTECTED]:
0057510:[EMAIL PROTECTED]
[voipexten]
authuser=0057510
username=0057510
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Gilberto Nunes wrote:
A Monday 14 January 2008 16:25:15, Steve Johnson escreveu:
Yeah! I'm just do this right now!
But I want more!
How can I create some extension to call to user, and pass the information
about
new voicemail message?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ruben Zamora wrote:
I have a little problem with mi outside calls. I Have Echo. It`s randomly.
I install codec G729 and I have Grandstream GXP2020.
Asterisk 1.4.9,Unicall 0.0.5pre1,Zaptel 1.4.5
I have a E1 and 30
Yes, this prompt will shows up on SIP 2.2.2 as well.
I never had any issues with this though, it will clear up after next
registration of phone.
I just downloaded SIP 3.0 and have not got a chance to check and see if it
happens with this firmware as well.
On Jan 22, 2008 2:53 PM, Steve Johnson
On Jan 22, 2008, at 10:23 AM, Philipp Kempgen wrote:
Andre Herrlich wrote:
any one advise a good, strong and free softphone that can work
with SIP
or/and IAX lines and supports attended transfer ?
IMHO there are no good softphones - at least not for
Mac OS X and I think that is true
hi,
I have used asterisk 1.2.12.1 and using linux 4 enterprise edition.
Bhrugu Mehta
On Jan 22, 2008 11:33 AM, ram [EMAIL PROTECTED] wrote:
On Jan 22, 2008 9:36 AM, Bhrugu Mehta [EMAIL PROTECTED] wrote:
hi, all
I set up asterisk with 5 to 6 agent . in these all are going well. but
when
--- zoa [EMAIL PROTECTED] wrote:
Hello,
Have you tried our Zoiper softphone yet
(www.zoiper.com) - new version
scheduled for in a couple of days ? If so, can you
send me any remarks
of list so that we can keep those things in mind for
future versions ?
Attended transfer is not
Monday 21 January 2008 coco wrote:
I have a problem with my asterisk server, I want to disable the call on
hold function when flash hook is pressed.(actually to fully disable it for
the users connected to the box) It does call on hold when I use the
asterisk as a rtp proxy, when it does
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