Yeah thanks a lot,
It worked that way...
Oguzhan Kayhan schreef:
Oguzhan Kayhan wrote:
I want to change it to E1 instead of T1.
here comes the problem.
If it's anything like the older cards, there is a jumper on the card
that sets it to T1/E1
Doug
Yes,
I just noticed the jumper on
2009/2/26 Brandon B. bran...@brellsystems.com
I've got a question about codec_dahdi witrh a system running Asterisk
1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to
route calls between different PRI connections, so no transcoding between
codecs is happening as far
On Thu, Feb 26, 2009 at 08:41:25AM +0200, Oguzhan Kayhan wrote:
Oguzhan Kayhan wrote:
I want to change it to E1 instead of T1.
here comes the problem.
If it's anything like the older cards, there is a jumper on the card
that sets it to T1/E1
Doug
Yes,
I just noticed the
Hello,
We need a US Dialing Plan to use it with Asterisk. We need 3 - 4 Channels
with it. Please suggest some good voice quality service.
Regards,
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email:
On Thu, Feb 26, 2009 at 08:41:25AM +0200, Oguzhan Kayhan wrote:
Oguzhan Kayhan wrote:
I want to change it to E1 instead of T1.
here comes the problem.
If it's anything like the older cards, there is a jumper on the card
that sets it to T1/E1
Doug
Yes,
I just noticed the
Tiago Durante wrote:
Hi all,
I don't know if its the right place to ask, but... Does any one have
the asterisk-stat-v2 running with PHP5?
Tks!
# php --version
PHP 5.2.0-8+etch13 (cli) (built: Oct 2 2008 08:26:18)
Copyright (c) 1997-2006 The PHP Group
Zend Engine v2.2.0, Copyright
amit mehta wrote:
Hello Users,
Is anyone aware about a solution to call incoming number and dictate the
files by using Dictate feature of Asterisk used for Medical
Transcription industry.
I guess nobody will read your email as you:
1. hijacked a thread
Hello Members,
Sorry for hijacking the earlier thread and asking the question last time.
Is anyone aware about a solution to call incoming number and dictate
the files by using Dictate feature of Asterisk used for Medical
Transcription industry.
Thanks Regards,
Amit Mehta
Does anyone have ideas on how to debug this?
I connected a different and old fax machine and everything works fine.
But for sure the other fax machine works fine too if I connect it to an
analog line from the telco.
Si it some incompatibility between my setup (asterisk, xorcom) and
the fax
On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote:
Hello,
I have connected a fax machine to a xorcom fxs port (signalling=fxo_ks).
But when I try to dial (send a fax) the number that I receive in
asterisk is wrong. Quite often a few digits are wrong but sometime is
correct. It
On Thu, Feb 26, 2009 at 11:25 AM, Hans Konings h...@konings.nu wrote:
I've tried with a different server (HPdl320g5p) and the card is detected in
this but the cards generate NMI errors on many bootups.
Does anybody have this combination of hardware working? Or can anybody
think of something
Hi
I'm having problems getting the TE420 working in HP DL380G5 servers.
The cards don't seem to be detected 100% by the BIOS. With two cards in the
server they are never detected.
things I've tried:
1 Update firmware to latest (P56) for the server
2 change irq settings
3 disable all onboard
Hello,
I use asterisk to to IAX2 trunking between London POP Reunion Island pop.
I would like to know if it's possible to do a kind of call-limit (i.e.
restrict to XX) channels but on a per dialcode and / or destination basis.
For example:
[trunk]
; reunion proper, i want to send no more than
Hans Konings schreef:
Hi
I'm having problems getting the TE420 working in HP DL380G5 servers.
The cards don't seem to be detected 100% by the BIOS. With two cards
in the server they are never detected.
things I've tried:
1 Update firmware to latest (P56) for the server
2 change irq
I have no clue about IAX, but if IAX does not support it you can program
it yourself using the GROUP and GROUPCOUNT functions.
regards
klaus
Jean-Michel Hiver wrote:
Hello,
I use asterisk to to IAX2 trunking between London POP Reunion Island
pop. I would like to know if it's possible to
Yes,
its the dialed number.
I have asterisk 1.4.21.1 and zaptel-1.4.11. Reading your question I am
pretty sure you advise me to update.
Loic
On Thu, 2009-02-26 at 12:44 +0200, Tzafrir Cohen wrote:
On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote:
Hello,
I have connected a fax
Look it up at www.localcallingguide.com. The name on record there may be
the LECs name -- then you need to find a SIP provider or DID handler that
has a business relationship with that LEC. For example, in the state of
Iowa Vonage obtained (at least some of) it's numbering resources from
As always, heed should be taken that number pooling and LNP have made
the idea that someone owns a rate center increasingly meaningless.
See the first or second question on our VoIP FAQ for more information
@ www.evaristesys.com. It is related. I would paste direct URL but am
not at
On Thu, Feb 26, 2009 at 11:49 AM, stoffell stoff...@gmail.com wrote:
On Thu, Feb 26, 2009 at 11:25 AM, Hans Konings h...@konings.nu wrote:
We have it working on a dl320, several ML350's, ml310, but never tried on a
dl380 yet.
We had serious issues in the past when iLO was enabled on a
On Wed, 25 Feb 2009, Klaus Darilion wrote:
I supsect that the incoming request is tried to match against a peer -
based on IP:port. Thus, it will always match the same peer, regardsless
if the call is incoming from account1 or account2.
Try using the same context in both peer definitions and
Ok, then you can contact one of the myriad of providers to see if you
can get a block of numbers. Keep in mind, there will be differences in
quality and reliability depending on which provider you use and on
whether or not you use PRI or pure VoIP. Since you already have the
outbound
According to Wikipedia no law was actually ever passed. I don't know if
there is anything that falls under the FCC rules. In any event it would
be unethical and evidence of fraudulent intent if one was trying to
defraud someone in the process of doing so.
Jason Aarons (US) wrote:
Any idea
Hi,
Changing the line bellow helped to get incoming calls but I add to remove
secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth
required challenges).
If someone could enable secret and still get incoming calls (in any
SmartWare 5.X), please, do not hesitate to share here
Hans Konings escribió:
Hi
I'm having problems getting the TE420 working in HP DL380G5 servers.
The cards don't seem to be detected 100% by the BIOS. With two cards
in the server they are never detected.
Did you test the TE420 card on another server? It may be a defective card...
--
Ing.
Hi, using EAGI variables like
agi_request
agi_channel
agi_language
agi_type
agi_uniqueid
agi_callerid
agi_dnid
agi_rdnis
agi_context
agi_extension
agi_priority
agi_enhanced
agi_accountcode
I get a lot of data about a call, but I need to obtain P-Asserted-Identity
value from a SIP call. Are
Hello! I?ve connected an avaya PABX with an asterisk box through h323, all
calls from Avaya are sended to the asterisk. What I need is send to the
AVAYA PABX a congestion tone when Zap channels are full. How I do it?Thanks
for any idea!
Gustavo A. González
Dto. de Infraestructura
Hi,
I am looking for a good ATA recommendation, ideally something:
1) with one FXS and one LAN port (so it's as inexpensive as possible)
2) That can be provisioning using FTP (configuration and firmware upon
reboot, ideally remote reboot from a sip notify)
3) Supports T.38
Nice
hi
you should first solve this
Warning [2630]: config.c:768 process_text_line: Unknown Directive at
line 231 of /etc/asterisk/../zaptel.conf
check what do you have in the line 231 of your zaptel.conf file.
David
2009/2/26 Wye-khe Kwok wye-khe.k...@biosjp.com
Hi everyone!
I'm quite a newbie
I used the Patton M-ATA a year or so ago and it was a piece of junk.
T.38 didn't work and there was no way to troubleshoot why it didn't
work. Also, the web interface was horrible.
Mike wrote:
Hi,
I am looking for a good ATA recommendation, ideally something:
1) with one FXS
Based on this page -
http://blog.herbertm.ca/2007/09/03/extracting-dids-from-the-sip-header
You could put this in your dialplan
[ext-did-custom]
exten = s,1,Set(ASSERT=${SIP_HEADER(Call-ID)})
exten = s,2,AGI(xxx.pl,${ASSERT})
_
From: asterisk-users-boun...@lists.digium.com
Thanks for the comment, it's good to know.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of pe...@networkoblivion.com
Sent: Thursday, February 26, 2009 9:16
To: Asterisk Users Mailing List -
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has had this problem.
When calling from an outside line, and entering the digits during the
read() part of my dialplan, it's accepting some of the digits twice,
though it's only keyed in
Klaus Darilion klaus.mailingli...@pernau.at writes:
What about a config option
gototriggersinvalid=yes (default=no)
in extensions.conf for users which are using this feature?
Please, no more options. There are way too many options already.
Since I personally believe the use of the special
Btw, I'm using Asterisk SVN-branch-1.4-r178640
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has had this problem.
When calling from an outside line, and entering the digits during the
read() part of my
Can you access channel functions from EAGI?
SIP_HEADER(P-Asserted-Identity) contains what you need.
It would be nice if Asterisk supported it natively and forgot about
the deprecated Remote-Party-ID header.
/Benny
___
-- Bandwidth and Colocation
Mike l...@virtutel.ca writes:
2) That can be provisioning using FTP (configuration and firmware upon
reboot, ideally remote reboot from a sip notify)
What's wrong with HTTP?
I have been recommended the PAP2T in the past, and although I have used it
and sort of liked it, it wasn't possible
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has had this problem.
When calling from an outside line, and entering the digits during the
read() part of my dialplan, it's accepting some of the digits twice,
On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote:
hi
you should first solve this
Warning [2630]: config.c:768 process_text_line: Unknown Directive at
line 231 of /etc/asterisk/../zaptel.conf
In zaptel.conf it is perfectly legal to have lines beginning with a '#'.
With Asterisk
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has
Hi,
I am trying to get * log to mssql server. I have odbc and freetds
configured, but my insert query is missing calldate which is a NOT
NULL field in database schema.
cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'. CDR failed:
INSERT INTO cdr
On Thu, 26 Feb 2009, Mike wrote:
Hi,
I am looking for a good ATA recommendation, ideally something:
1) with one FXS and one LAN port (so it's as inexpensive as possible)
2) That can be provisioning using FTP (configuration and firmware upon
reboot, ideally remote reboot from a sip
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
Sorry for wasting your time.
--
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 *
On Thursday 26 February 2009 08:39:29 Benny Amorsen wrote:
Can you access channel functions from EAGI?
SIP_HEADER(P-Asserted-Identity) contains what you need.
It would be nice if Asterisk supported it natively and forgot about
the deprecated Remote-Party-ID header.
Of course you can.
GET
amit mehta wrote:
Hello Members,
Sorry for hijacking the earlier thread and asking the question last time.
Is anyone aware about a solution to call incoming number and dictate
the files by using Dictate feature of Asterisk used for Medical
Transcription industry.
Thanks Regards,
What is illegal is to set caller-id to a fraudulent value such that the
person on the other end will not be able to correctly identify the
originator of the call.
I don't know if there is anything that falls under the FCC rules. In any
event it
would be unethical and evidence of
Not at all.
In fact, I found that relaxdtmf=yes is now available for sip.conf as of
1.4 as well.
However, that didn't resolve the problem.
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with
Thanks to all, I'll try it!
On Thu, Feb 26, 2009 at 12:25 PM, Danny Nicholas da...@debsinc.com wrote:
Based on this page -
http://blog.herbertm.ca/2007/09/03/extracting-dids-from-the-sip-header
You could put this in your dialplan
[ext-did-custom]
exten =
Hi Tiago.
I've it working on PHP 5.2.6 but only after having modified the php.ini
default configuration keys:
zend.ze1_compatibility_mode = Off
short_open_tag = Off
setting together to On and restarting apache forces PHP5 to behave like
PHP 4.x version.
regards,
Marco Signorini
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and
Hi list,
I wonder if anyone is able to offer any [polite ;-)] words of wisdom??
I'm having the same problem too, with random DTMF heard during ISDN calls. Am
running Asterisk 1.4.22 and mISDN 1.1.8. (kernel 2.6.18 CentOS)
Symptoms are on incoming and out outgoing ISDN calls,
Yea, I tried that too. I have it: dtmfmode=rfc2833
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert
Hi Brent,
Thanks for the help.
I am working to achieve the middle solution.
The scenario that i am trying to achieve :
Customer calls in the tollfree number:
Enter the ID and press#
Pwd: #
Press * to begin dictate,4 to pause and 2 to start again.
PRess * to end and start new
Upto this has been
Just beware that most PAP2s on places like E-Bay are from Vonage. They lock
the things up quite seriously. There are procedures out there to unlock them
but it requires stuff like setting up an isolated LAN with a DNS server and FTP
server and a special file. If someone would make a live
On Thu, 26 Feb 2009, Wilton Helm wrote:
Just beware that most PAP2s on places like E-Bay are from Vonage. They
lock the things up quite seriously. There are procedures out there to
unlock them but it requires stuff like setting up an isolated LAN with a
DNS server and FTP server and a
Hey all, I have a couple of questions.
1) What is the maximum amount of registrations and ongoing
calls you've been able to achieve on your Asterisk systems.
Please do not respond with marketing hyperbole. I'm looking
for real world implementation in the thousands range. For
instance, max I
Marco Signorini wrote:
Hi Tiago.
I've it working on PHP 5.2.6 but only after having modified the php.ini
default configuration keys:
zend.ze1_compatibility_mode = Off
short_open_tag = Off
Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is
set to On and it is working.
Hi,
With 0.0.6pre3:
# ./build.sh
CMake Warning (dev) in CMakeLists.txt:
No cmake_minimum_required command is present. A line of code such as
cmake_minimum_required(VERSION 2.6)
should be added at the top of the file. The version specified may be
lower
if you wish to support older
I must add I tried spandsp0.0.6xxx as a warning message advised me to do so
(using 0.0.4 would be ok for me but current trunk doesn't allow this
anymore, it seems).
2009/2/26 Olivier oza-4...@myamail.com
Hi,
With 0.0.6pre3:
# ./build.sh
CMake Warning (dev) in CMakeLists.txt:
No
2009/2/26 pe...@networkoblivion.com pe...@networkoblivion.com
I used the Patton M-ATA a year or so ago and it was a piece of junk.
T.38 didn't work and there was no way to troubleshoot why it didn't
work.
Have you tried again recently ?
I've just tried its T.38 capabilities and I'm not
2009/2/26 Mike l...@virtutel.ca
a) PoE powered
Unfortunately, I don't think any PoE powered exists, AFAIK.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Steve:
=
Thanks for the info on the agi debug command. We'll see what information
we can garner with that. Thanks also for the advanced logging info.
Unfortunately, we are pretty aware of how AGI works (at least at the
level that you explained it). Thanks for the
Dear All,
I have created an inbound context in SIP .conf that forward incoming call to
opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
voice codec to INVITE packet...It just contains T.38
An update here. Yesterday's problem has been solved (partly). After
looking closer at the results of my perl script, as well as looking at
the documentation, I realized that we were leaving the escape character
argument off of the STREAM FILE command in the mono application. After
appending to
On Thu, Feb 26, 2009 at 6:08 PM, Simon Dixey simon_...@hotmail.co.ukwrote:
I wonder if anyone is able to offer any [polite ;-)] words of wisdom??
I can be polite, I'm not sure about the wisdom .. :-)
DTMF threshold in misdn-init look high doesn't it... I'm not entirely sure
what it should
On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos paulo.r.san...@sapo.pt wrote:
Marco Signorini wrote:
Hi Tiago.
I've it working on PHP 5.2.6 but only after having modified the php.ini
default configuration keys:
zend.ze1_compatibility_mode = Off
short_open_tag = Off
Though my
hi all,
I'm have a bit of a hard time with some segfaults on running 1.4.23.1 and
mISDN 1.1.8. I already enabled DONT_OPTIMIZE and DEBUG_THREADS in
asterisk so I can now generate a bt. I did that (following the instructions
on voip-info) but I'm not sure how to read' the output now.
By looking
I turned on DTMF debugging. It looks like the extra digits coming in are
less than the minimum duration of 100ms
Anyone know how to force that minimum duration?
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1'
received on SIP/carrier-c4022740
[Feb 26 12:15:07]
sorry but how do you know the warning is from an # ?
he only post this from zaptel.conf
span = 1,1,0,esf,b8zs
bchan = 1-23
dchan = 24
loadzone = jp
defaultzone = jp
David
2009/2/26 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote:
hi
you
Hi,
We're moving ahead on this, there are some interesting files to check
out thanks to davevg's perl skills. If you find a way to write an
interesting application to run on a Polycom, you can win a Polycom IP
450. Even if you don't care about that, you may be interested in
knowing more about
Robert Broyles wrote:
I turned on DTMF debugging. It looks like the extra digits coming in
are less than the minimum duration of 100ms
Anyone know how to force that minimum duration?
[Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin
'1' received on SIP/carrier-c4022740
[Feb
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?
I've read lots of threads going back to 2007 and I'm in the general
option that kvm is the way
Nope, I gave up and haven't gone back. Not worth the hassle. We use
the SPA2100/SPA2102 and they work great for providing analog/fax lines
as they support T.38.
Olivier wrote:
2009/2/26 pe...@networkoblivion.com mailto:pe...@networkoblivion.com
pe...@networkoblivion.com
On Thu, 26 Feb 2009, Douglas Mortensen wrote:
New problem:
==
Although we see the following in our logs / asterisk console: [Feb 26
11:11:05] VERBOSE[9824] logger.c: -- Playing 'filename'
(escape_digits=) (sample_offset 0)
All that the caller hears is a very brief click. And
At the top of my /etc/dahdi/system.conf file is this line:
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009
-- do not hand edit
OK, so how do I adjust the timing source and LBO numbers, and echo
cancellers if I'm not supposed to edit this file?
Brandon.
2009/2/25 stoffell stoff...@gmail.com
Hi all,
I wanted to switch from my current setup (mISDN) to the native dahdi with
b410p support (wcb4xp). All works fine for normal phone calls but not for
faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual
bad stuff (HDLC frame
At the top of my /etc/dahdi/system.conf file is this line:
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 --
do not hand edit
OK, so how do I adjust the timing source and LBO numbers, and echo cancellers
if I'm not supposed to edit this file?
Well, if you hand edit
On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote:
At the top of my /etc/dahdi/system.conf file is this line:
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009
-- do not hand edit
OK, so how do I adjust the timing source and LBO numbers, and echo
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Brandon B. wrote:
At the top of my /etc/dahdi/system.conf file is this line:
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10
2009 -- do not hand edit
OK, so how do I adjust the timing source and LBO numbers, and echo
You need canreinvite=no in the config for your sip phone and the
veracity connection, otherwise Asterisk will just mediate the call setup
then try to allow the sip phone and veracity to talk directly to one
another.
Jim Dickenson wrote:
I have a SIP phone at home behind a NAT router
Actually you want to use the one with -N/A at the end.
Thanks for the correction. My memory failed me. But definitely I want people
to know that not every PAP2T is useful to them so they don't get burned. IMO
Vonage (and others) should not be locking them the way they do (and Linksys
On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote:
sorry but how do you know the warning is from an # ?
'Directive' is something that begins with a '#'.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
On Thu, 26 Feb 2009, Wilton Helm wrote:
Actually you want to use the one with -N/A at the end.
Thanks for the correction. My memory failed me. But definitely I want
people to know that not every PAP2T is useful to them so they don't get
burned. IMO Vonage (and others) should not be
The problem turned out to be a firewall issue. If one makes a call out the
PRI line * sends the ring audio to the sip phone. This opened a hole in the
firewall for the return traffic so things worked. If I make the call from my
office when the call was answered and the caller started talking and
On Wed, 2009-02-25 at 19:14 -0200, David fire wrote:
please keep us informed about it.
David
2009/2/25 Kristian Kielhofner kristian.kielhof...@gmail.com
Hello everyone,
I just ordered one of these:
On Thu, Feb 26, 2009 at 10:35:31PM +0100, Hans Witvliet wrote:
Like to know if the power plug is only suitable for USA?
But can also take 220V, according to specs.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
At 01:25 PM 2/26/2009, you wrote:
I don't mind them
providing autoconfiguration, as many of their customers would be lost
otherwise, but blocking factory reset and other tricks they play is IMO
morally wrong, particularly since the customer paid for the
ATA.
I have a locked ATA here and I didn't
On Thu, 26 Feb 2009, Ira wrote:
I have a locked ATA here and I didn't pay for it. So while I'm not happy
it's locked and I can't use it, I can't complain that much as it was
free.
It may be more accurate to say I have a locked ATA that was paid for by
my provider charging me a little bit
Hello,
Some of my users have phones lacking a DND button. I need to provide
an extension they can dial that will put them in DND, i.e. tell the
server not to send them any calls until they get off the DND.
I've researched it for almost 3 days now and tried a range of
configurations. I'm
thanks
2009/2/26 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote:
sorry but how do you know the warning is from an # ?
'Directive' is something that begins with a '#'.
--
Tzafrir Cohen
icq#16849755
Hi,
I'm using a Cisco IP Communicator 7.0 enable with SIP, but I can't to
configure it to add the lines because I can't to Select the options, for
example: the Name, Authentication name...
Somebody help me??
Thank you!!!
___
--
Gavin Henry wrote:
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?
I've read lots of threads going back to 2007 and I'm in the general
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
smime.p7s
Description: S/MIME Cryptographic Signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
In a nut shell the CHANNEL variable is just that variable. It has a call leg id
attached to it so if that is what you are storing it will change everytime you
create a new channel.
For example if I place a call Thru SIP channel polycom1 the channel is:
SIP/polycom1-23a3bc, You could look at
Digium suspects these cards have a problem, so they are being rma'd.
http://www.digium.com/en/docs/tech_bulletins/20081113.php
Thanks for the input.
On Thu, Feb 26, 2009 at 2:34 PM, Miguel Molina mmol...@millenium.com.cowrote:
Hans Konings escribió:
Hi
I'm having problems getting the
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