Thanks all :)
Appreciate it.
On Feb 1, 2008 12:04 PM, Ex Vito [EMAIL PROTECTED] wrote:
I've struggled with this recently. In short:
- Observed behaviour is expected as of asterisk 1.2 and later,
as previously described by Mojo
- If you want to get the caller id for the channel
I am getting quite a lot of these notices:
Feb 1 10:22:01 NOTICE[2487] cdr.c: CDR on channel
'mISDN/3-2' not posted
Feb 1 10:22:01 NOTICE[2487] cdr.c: CDR on channel
'mISDN/3-2' lacks end
Feb 1 10:22:06 NOTICE[2471] cdr.c: CDR on channel
'mISDN/3-1' not posted
Feb 1 10:22:06 NOTICE[2471]
Hi,
I have the following problem that when someone connects to my conference and
is the only member
music on hold is played just for one second or less and then stops:
[Feb 1 10:38:46] -- Started music on hold, class 'default', on channel
'SIP/sip.touk.pl-0083dad0'
[Feb 1 10:38:46] --
You can use account code and userfield. You can set userfield to anything
you want in the dialplan.
On Feb 1, 2008 3:31 PM, Doug Lytle [EMAIL PROTECTED] wrote:
Paul Hales wrote:
Anyone have any ideas? What can I use to carry a variable over into
'h'??
Lets see what you have so far.
Paul Hales wrote:
Anyone have any ideas? What can I use to carry a variable over into
'h'??
Lets see what you have so far.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote:
Hi,
I have the following problem that when someone connects to my conference and
is the only member
music on hold is played just for one second or less and then stops:
[Feb 1 10:38:46] -- Started music on hold, class 'default', on channel
On 2/1/08, Paul Hales [EMAIL PROTECTED] wrote:
I need to carry a variable over into the 'h' priority - so I can go back
and clean up DB entries in a mysql database (time of call and so on)
I tried using UNIQUEID but it seems that 'h' generates a new one.
Anyone have any ideas? What can I
Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel
1.4.5.1. If we were to update or recompile Asterisk, would we need to
do anything with Unicall or Zaptel?
Thanks in advance
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Hello,
I want that, when call is answered , callee and calling would hear
different prompts and after promts the calls would be bridged. I've tried
this situation:
exten = s,1,Set(LIMIT_CONNECT_FILE=hello-world)
exten =
[EMAIL PROTECTED] wrote:
Am I doing something wrong? What I should do to get ooh323.conf
cp asterisk-ooh323c/h323.conf.sample /etc/asterisk/ooh323.conf
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
___
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2008/2/1, Giedrius Augys [EMAIL PROTECTED]:
Hello,
I want that, when call is answered , callee and calling would hear
different prompts and after promts the calls would be bridged. I've tried
this situation:
exten = s,1,Set(LIMIT_CONNECT_FILE=hello-world)
exten =
On Fri, 2008-02-01 at 15:01 +0100, Alberto Pastore wrote:
Olivier ha scritto:
Hi,
Does such card exist ?
It seems all existing models are designed for PCI buses.
Regards
yes.
On Feb 1, 2008 12:07 PM, Russell Bryant [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
Hi,
The server log shows the following message.
[Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available
Does it
i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be
more stable than Fedora core Linux or it makes no significant difference
_
Express yourself instantly with MSN Messenger! Download today it's FREE!
Olivier ha scritto:
Hi,
Does such card exist ?
It seems all existing models are designed for PCI buses.
Regards
___
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Hi,
Does such card exist ?
It seems all existing models are designed for PCI buses.
Regards
___
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On Thursday 31 January 2008 11:52:09 pm Ian wrote:
Sorry for taking so long to reply,
This email got lost in translation, again.
Ian
Ian said the following on 30-Jan-08 03:57 PM
Thaks for the speedy reply
Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:
On Wed, Jan 30, 2008
love U.all wrote:
i wanna build a production Asterisk box ,will RedHat Linux Enterprise
Server be more stable than Fedora core Linux or it makes no
significant difference
I started out running Fedora, but I have migrated away from it for a few
reasons. Fedora has a very short life cycle
Matthew J. Roth a écrit :
[...]
I settled on using an empty TDM400P as a timing source, because it is a
simple solution that just works. This may still be your best bet, but
I'll defer judgment on that to the list because Asterisk has evolved
quite a bit since I made that decision.
This
I have,
I have ztdummy module loaded in the kernel
On Feb 1, 2008 11:59 AM, Gordon Henderson [EMAIL PROTECTED]
wrote:
On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote:
Hi,
I have the following problem that when someone connects to my conference
and
is the only member
music on hold is
Thanks Tzafir, but this functionality needs sombody answer the call.
I need to do this automatically.
On Jan 22, 2008 4:10 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
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?
We are using Fedora because that is what the company we got our system
from recommended.
If I was doing the system myself I would throw in my vote for CentOS.
I am using it for a database server and I have had no problems with it
at all. It is about as stable and secure of Linux distro as I have
Anyone aware of any SIP softphones that might virtualize well with Citrix
presentation server? I suspect I know the answer already as I have been
researching softphones that work with Cisco CallManager that can be
virtualized if you will with Citrix and have come to learn that its not
something
Mark, you are confusing terms here.
You DO NOT have Unicall 1.4.9-0.1, libunicall does not even use that
convention for its versions. What you have is astunicall-1.4.9-01.
AstUnicall is just a package with patches and proper versions of
spandsp, libsupertone, libunicall, libmfcr2, zaptel and
Anyone using Asterisk in a Call Center environment? And more importantly is
anyone supporting home based remote call center agents with an Asterisk
backend?
My experience with Asterisk is limited, however I have set it up and
installed it previously and had it working for home usage and for
That sounds like a CANCEL message being improperly routed somewhere
along the line.
Nothing you can do with the config to fix that.
N.
Paul Madley wrote:
Hi,
Does anyone of you has a working configuration with SNOM phones that are
able to pickup a call from a flasing LED?
Thank you Moisés, we are indeed going from 1.4.9 to 1.4.17, we will backup
channels/chan_unicall.c and the Makefile entries of channels/Makefile and do
our upgrade to .17.
You are indeed correct on the Unicall, we have astunicall-1.4.9-0.1, I thought
it was the same thing. Now I know better
Download for Pentium4
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Wednesday, January 30, 2008 10:55 PM
To: asterisk-users@lists.digium.com
Administrator TOOTAI wrote:
This is not true if you're using B410P cards. We always face timing
problem as we can't -Asterisk stability issues- add X100P or TDM400P
with those cards
Daniel,
I thought that using an empty TDM400P as a timing source may no longer
be the best solution due to
Hi,
Does anyone of you has a working configuration with SNOM phones that are
able to pickup a call from a flasing LED?
Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and
therefore I don't think any config changes will fix it. We've been told to
roll back to our
Matthew J. Roth wrote:
love U.all wrote:
i wanna build a production Asterisk box ,will RedHat Linux Enterprise
Server be more stable than Fedora core Linux or it makes no
significant difference
I started out running Fedora, but I have migrated away from it for a few
reasons. Fedora
Hello,
For such cases we usually suggest to put 2 boxes in your infrastructure:
1. Main billing gateway - where all PBX'es are connected (all client's remote
PBX'es and your Local PBX)
2. Local PBX - where user's without PBX'es are connected
Then user connects in following way:
User - Local
Ok, I have made some progress debugging this. I dont believe it has
anything to do with asterisk or my phone. Rather I think it is an
issues with STUN and/or my Linksys router at home.
The phones I am testing all sit behind a NAT'd firewall, your basic
Linksys router for the Home DSL user.
Hi List;
Is it possible to do configuration at the user context
to let him use codec1 if destination support and if
not, then use codec 2?
For example, to let user1 use codec g729 if he needs
to call user2 and user2 support g729, and if user2
does not support g729 then use g711 (alaw or ulaw),
My suggestion - use the distro which you know best.
We use Debian (200+ installations). It works stable for us because we know how
to achieve it.
Others use Fedora/Centos - because they are experts in these systems.
Stability and performance of the system does not depend on the distro - only
On Feb 1, 2008 10:33 AM, d4rk f1br [EMAIL PROTECTED] wrote:
Anyone using Asterisk in a Call Center environment? And more importantly is
anyone supporting home based remote call center agents with an Asterisk
backend?
My experience with Asterisk is limited, however I have set it up and
On Feb 1, 2008 10:33 AM, d4rk f1br [EMAIL PROTECTED] wrote:
Anyone using Asterisk in a Call Center environment? And more importantly is
anyone supporting home based remote call center agents with an Asterisk
backend?
My experience with Asterisk is limited, however I have set it up and
Hello Steve,
You are right on track and this is also what we have done with pretty good
results.
Of course now with Flex/Air there are a number of ways to enhance the
service for the
Customer/Agent
Ed
Mail: edpimentl[at]gmail.com
Voip: edpimentl [SKype | GoogleTalk ]
http://agileoss.com
Trust me, I don't WANT you to look at my code, it's butt-ugly! lol, lust
kidding... -- but at http://www.astsee.com/ you can download the source
code to my AstSee project -- it may provide some insight into what needs
to be (or CAN be) gleaned from asterisk. I struggled with all this a
year
On Fri, 1 Feb 2008, Milton Calnek wrote:
Matthew J. Roth wrote:
love U.all wrote:
i wanna build a production Asterisk box ,will RedHat Linux Enterprise
Server be more stable than Fedora core Linux or it makes no
significant difference
I started out running Fedora, but I have migrated
Hello All:
Does the Asterisk support to insert an off the board transcoder for a call?
Thanks,
Charles
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
http://www.digium.com/en/products/voice/tc400b.php
Simon Elliston Ball
[EMAIL PROTECTED]
On 1 Feb 2008, at 17:29, Charles Feng wrote:
Hello All:
Does the Asterisk support to insert an off the board transcoder
for a call?
Thanks,
Charles
Never miss a thing. Make Yahoo your
On Fri, Feb 01, 2008 at 09:58:28AM -0600, Milton Calnek wrote:
Matthew J. Roth wrote:
love U.all wrote:
i wanna build a production Asterisk box ,will RedHat Linux Enterprise
Server be more stable than Fedora core Linux or it makes no
significant difference
I started out running
I am having a problem with DTMF when sending calls through Teliax
(SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the
most part it is working. The problem always happens when a user is
trying to call a conference system. They simply cannot get into the
conference
I cannot think of a single reason to use Fedora for a production anything
when there are alternatives like CentOS. Fedora is bleeding edge stuff and
constantly changing.
-Original Message-
From: Matthew J. Roth [mailto:[EMAIL PROTECTED]
Sent: Friday, February 01, 2008 7:39 AM
To:
Hello,
I have 2 asterisk servers that are not working well together. One is
acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX
devices. And the other is acting like my sip gateway (PBX02) to
various providers. They are both on a private network and should be
trusting
There a realtime LDAP driver now in 1.6beta2
On 23/01/2008, Cavalera Claudio Luigi [EMAIL PROTECTED] wrote:
Hello,
I've found this information about asterisk and LDAP:
http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
which can be out of date.
I'm trying this
Matthew J. Roth wrote:
love U.all wrote:
i wanna build a production Asterisk box ,will RedHat Linux Enterprise
Server be more stable than Fedora core Linux or it makes no
significant difference
I started out running Fedora, but I have migrated away from it for a few
reasons. Fedora
Just noticed this today:
Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo
Cancellation Modulehttp://www.voipsupply.com/product_info.php?products_id=3352
It's about time Digium got on the ball and made PCI-e cards. What are
people's experiences with this card? Anyone
Hi all,
What format is the LastCall variable of QueueMember event? I'm looking at:
1201897536 for instance.
Thanks !
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling encyclopedias door to
door.
___
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On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote:
What format is the LastCall variable of QueueMember event? I'm looking at:
1201897536 for instance.
Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I
recall.
--
Jared Smith
Community Relations Manager
Digium, Inc.
Matt wrote:
Just noticed this today:
Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based
Echo Cancellation Module
http://www.voipsupply.com/product_info.php?products_id=3352
It's about time Digium got on the ball and made PCI-e cards. What are
people's experiences
There is an option you might consider (if you are starting from scratch).
Don't use citrix. Write a web app.
Then embed a softphone in that web app.
Tim.
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asterisk-users
- Original Message
From: Mindaugas Kezys [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 1 February, 2008 4:04:30 PM
Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
For
such
cases
we
Jason.. great! How long have those analog cards been out? I don't see
them on my suppliers list.
Digium already makes PCI Express analog cards - AEX800 and AEX2400.
--
Jason Parker
Digium
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I'm working for zoiper.com and i'm willing to help out with ours when
needed.
Zoa
d4rk f1br wrote:
Anyone aware of any SIP softphones that might virtualize well with
Citrix presentation server? I suspect I know the answer already as I
have been researching softphones that work with
Jared Smith wrote:
On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote:
What format is the LastCall variable of QueueMember event? I'm looking at:
1201897536 for instance.
Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I
recall.
Thanks.
--
Warm Regards,
Lee
At 10:57 AM 2/1/2008, you wrote:
Any tweaks recommended for DTMF and Teliax?
try:
dtmfmode=auto
That's what mine is now after rfc2833 stopped working.
Ira
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shadowym [EMAIL PROTECTED] writes:
I cannot think of a single reason to use Fedora for a production anything
when there are alternatives like CentOS. Fedora is bleeding edge stuff and
constantly changing.
The advantage of Fedora is that it is very actively maintained -- and
asterisk is only
On Friday 01 February 2008 15:31, Matt wrote:
It's about time Digium got on the ball and made PCI-e cards. What are
people's experiences with this card? Anyone know if there are plans for a
PCI-e analog card for FXO use?
I have been using 220B's for about 6 months. I have about 20 of them
I am trying to setup SIP to SIP calling between Asterisk managed networks. I
want to make it so that people can call SIP:[EMAIL PROTECTED] and they
connect to my Asterisk and get my external IVR then they can dial my
extension or navigate extensions just like they would if they had called
using a
The client is travelling much of the time.
Is there some way that he can use Port 80 so
that the firewalls that he is behind won't
block the connection?
Any other hints or suggestions are very
welcome!
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Hi, all. I've used the perl/AGI interface, and... well, I found it kind
of hokey. Granted, this was in 1.2 days -- perhaps things have changed.
Regardless, I guess I have two questions:
1) Has the Perl/AGI binding improved since then?
2) Is there any chance of a real API for Perl?
Thanks much!
Carlos Chavez wrote:
I am having a problem with DTMF when sending calls through Teliax
(SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the
most part it is working. The problem always happens when a user is
trying to call a conference system. They simply cannot get
I have Asterisk 1.4 registering via IAX to another Asterisk machine.
How can I change the default registration timeout of 60s?
I need my Asterisk box to register every HOUR Anyone?
Editting source isn't an option.
Doug.
Hello list,
New to asterisk and to the list (although experienced in Unix/Linux
administration).
Short problem description:
--
I cannot get the Echo() application to run on any 32bit platform I can get my
hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup
Hello,
On our Polycom phones we can not activate the Buddy Watch feature.
When you add or edit a contact, the list ends at Auto Divert.I know it
is the end of the list b/c the down arrow on the right side of the screen
disappears when I get to Auto Divert.
When I add bw1/bw manually to the
On Feb 1, 2008 3:53 PM, Thermal Wetland [EMAIL PROTECTED] wrote:
Hello,
On our Polycom phones we can not activate the Buddy Watch feature.
When you add or edit a contact, the list ends at Auto Divert.I know
it is the end of the list b/c the down arrow on the right side of the screen
On Fri, Feb 01, 2008 at 05:01:56PM -0800, Yassen Damyanov wrote:
Hello list,
New to asterisk and to the list (although experienced in Unix/Linux
administration).
Short problem description:
--
I cannot get the Echo() application to run on any 32bit platform I can
Hi Alberto,
2008/2/1, Alberto Pastore [EMAIL PROTECTED]:
Olivier ha scritto:
Hi,
Does such card exist ?
It seems all existing models are designed for PCI buses.
Regards
Ken D'Ambrosio wrote:
Hi, all. I've used the perl/AGI interface, and... well, I found it kind
of hokey. Granted, this was in 1.2 days -- perhaps things have changed.
Regardless, I guess I have two questions:
1) Has the Perl/AGI binding improved since then?
2) Is there any chance of a real
Doug wrote:
The client is travelling much of the time.
Is there some way that he can use Port 80 so
that the firewalls that he is behind won't
block the connection?
Any other hints or suggestions are very
welcome!
I suppose the usual answers apply. You can have Asterisk bind to UDP
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