Dear all,
First off, I am very new to asterisk so forgive me if any of my
comments or questions seem trivial. Thanks to [this
post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/)
and [this post](http://www.davidvossel.com/?p=28), I have GV set up on
asterisk through
although I don't need the solution personally But would like to request you
that instead of posting forget it . if you post the solution to the
problem it will be more helpful.
In case some one else faces the same problem he can use your solution
Good luck
On Sun, Oct 24, 2010 at 7:10
Hi.
There is /doc/core-en_US.xml in asterisk 1.8 source tree. Is this file
generated from documentation comments of apps/app_*.c files?
And how this file can be used? How can I convert it to pdf/html in order to use
it as applications documentation?
--
Am Donnerstag, den 21.10.2010, 16:27 +0200 schrieb Karsten Wemheuer:
Hi,
I setup an asterisk system (version 1.8.0-rc5). While using a SIP only
environment I discovered a problem using blind transfer. The phones are
SNOM or Aastra and are using the SIP REFER Method.
The following is
I would probably do this through the AMI, it should spew out the info
you require. Timestamp when they entered the queue and timestamp when
they get answered.
On 10/25/2010 05:01 AM, Bruce B wrote:
Anything on this guys?
I am sure someone had the need to record the HOLD time or maybe it is
On Mon, Oct 25, 2010 at 12:50 AM, Anthony Messina amess...@messinet.com wrote:
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for
On Sat, Oct 23, 2010 at 02:07:27PM +0200, Olivier wrote:
Hi,
My set up is :
Asterisk with B410P in NT mode -cat5 straight cable
Another PBX in TE mode
Is the 100 Ohm terminator you can find on B410P boards, necessary when
connecting in NT mode to another PBX
On Sat, Oct 23, 2010 at 02:43:26PM +0200, Olivier wrote:
Hi,
How to properly re-configure dahdi, when for instance I want to change from
TE to NT mode ?
I'm planning the following operations :
/etc/init.d/asterisk stop
/etc/init.d/dahdi stop
rmmod dahdi
rm
Hi, I am using Asterisk 1.6.2.13 and have an issue but I'm not sure if
it's a bug or not. I am using the cdr_adaptive_odbc logging module and
writing my CDR records to a MS-SQL server. I need to log which end
hangs the call up and have placed the relevant
CDR(myfield)=caller/callee commands
On 10/23/2010 07:07 AM, Olivier wrote:
Hi,
My set up is :
Asterisk with B410P in NT mode -cat5 straight cable
Another PBX in TE mode
Is the 100 Ohm terminator you can find on B410P boards, necessary when
connecting in NT mode to another PBX (set in TE mode)
Sorry, thats right!!
I the nest email I will post here what I did in order to sove my problem!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Sun, 24 Oct 2010 23:59:27 -0700
From: shakeel.abbas@gmail.com
To: asterisk-users@lists.digium.com
On 10/23/2010 07:35 AM, ali raza wrote:
Hello
I am working on TDM2400p. I am having some problems like:
when i connect my analog phone with the card there is no dial tone, but
i can dial any extension... but after that i can't hear any voice from
my receiver i have used different phone sets
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH
Sent: Saturday, October 23, 2010 7:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] a2billing muting enter
Thanks for the feedback. I don't need the Queue times but rather putting ON
HOLD times. If you press the HOLD button on your SIP phone, Asterisk records
the event Music On HOLD Playing and that is recorded in
/var/log/asterisk/full. I want to harvest the ON HOLD time per phone SET.
Thanks
On
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Monday, October 25, 2010 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best way to recording the hold time for a
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Sunday, October 24, 2010 4:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Chan variables for peer
Hi all,
I used to
But I don't want to delete the file!
I just want to know where is the option for playing or disabling this
message???
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday, October 25,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH
Sent: Monday, October 25, 2010 4:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] a2billing muting enter
Thanks for the input.
Would I have to process each call through a specific dial-plan for the AMI
to be in charge of each call so that it can see the Unique ID of the channel
and the Hold event? Because that seems like a lot of work. If AMI (I have no
experience with it) allows me to open a socket
2010/10/25 Shaun Ruffell sruff...@digium.com
On 10/23/2010 07:07 AM, Olivier wrote:
Hi,
My set up is :
Asterisk with B410P in NT mode -cat5 straight cable
Another PBX in TE mode
Is the 100 Ohm terminator you can find on B410P boards, necessary when
On Monday 25 October 2010 8:53:19 am Danny Nicholas wrote:
#1 check the bug tracker
#2 you might have to change the syntax - it seems to me that setvar=id=123
is an accident waiting to happen (or maybe it did.) setvar=id=123 or
setvar = id=123 might be more appropriate.
Thank you. I should
2010/10/25 Tzafrir Cohen tzafrir.co...@xorcom.com
On Sat, Oct 23, 2010 at 02:43:26PM +0200, Olivier wrote:
Hi,
How to properly re-configure dahdi, when for instance I want to change
from
TE to NT mode ?
I'm planning the following operations :
/etc/init.d/asterisk stop
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Monday, October 25, 2010 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best way to recording the hold time for a
Hi,
My asterisk's version is 1.6.0.26. I've couple sip providers and I've
for new SIP provider I need define outbound proxy. Everything is ok in peer
section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I
need send SIP register messages also via outbound proxy. How to
Any news for this problem.
Who has this problem
--- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit :
De: ALAEDDINE abbech alasup...@yahoo.fr
Objet: saturation of bandwidth because of HANGUP
À: asterisk-users@lists.digium.com
Date: Jeudi 21 octobre 2010, 17h55
Hello,
I
Any news for this problem.
Who has this problem
Why you don't answer.
--- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit :
De: ALAEDDINE abbech alasup...@yahoo.fr
Objet: thousands Hangup per second /saturation of bandwidth
À: asterisk-users@lists.digium.com
Date: Jeudi
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ALAEDDINE
abbech
Sent: Monday, October 25, 2010 10:52 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re : saturation of bandwidth because of HANGUP
Any
Un-self-top-posting...
--- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit :
De: ALAEDDINE abbech alasup...@yahoo.fr
Objet: thousands Hangup per second /saturation of bandwidth
À: asterisk-users@lists.digium.com
Date: Jeudi 21 octobre 2010, 11h42
You didn't attach some debug output that shows some work, and you didn't
even tell us what asterisk version are you using, which scenario is on, etc.
Don't expect people to run and answer right away with an inmediate
solution to this.
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone
Hi Jigar,
I use visual dialplan too. Nice tool.
Here you can find some dial plan examples and tutorials that may help you:
codezone.apstel.com
Nile
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon
Sent: Monday, October 25, 2010 12:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dial plan help
Hi Jigar,
I use visual dialplan too.
Ok Thanks Guys.
Can you guyz suggest me upto which chapters orwhat are the chapters I should
cover for my requirement.
Because Its too long book :P
On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:*
Chapters 4, 5 and 6 is a good start.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote:
Ok Thanks Guys.
Can you guyz suggest me upto which chapters orwhat are the chapters I should
cover for my requirement.
Because Its
Hi Everyone,
Which paid or unpaid commercial plugin is available out there for Asterisk
that would do Outlook contacts pop-up that is proven to work great with MS
Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well
through the Outlook.
Thanks,
Bruce
--
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Monday, October 25, 2010 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Pop-up for MS Outlook 2007 recommended
Hi
Un-top-posting...
On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote:
Ok Thanks Guys.Can you guyz suggest me upto which chapters orwhat
are the chapters I should cover for my requirement.
Because Its too long book :P
On Mon, 25 Oct 2010, Zeeshan Zakaria
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, October 25, 2010 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan help
You may check these videos too:
http://www.youtube.com/watch?v=H1j5OrgL1og
http://www.youtube.com/watch?v=7kNYuqOrP3w
I find it useful, although I use visual dial plan rather than hand
coding the dial plan.
Either way you need to understand at least basics of asterisk dial plan
structure.
Great suggestion but unfortunately for this client a proven technology is
needed and we don't mind paying a bit for it so once the time is available
we might do this the way you suggested.
Thanks
On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote:
I think there is a new version of Outcall, the pop up was pretty good, but the
dialout wasn't ideal in Win 7 , and I believe thats fixed now with good
integration with 2007 and 2010
http://code.google.com/p/outcall/
You can buy commercial options from Biocom - who make Outcall
Hi,
When a VOIP user dials an external number, the calls are routed through our SIP
provider.
Is there a simple way to check whether the DDI exists locally before dialling
out to the sip provider?
Something like GotoIfExists(5551...@incoming_calls)
Currently, I'm paying for all calls,
On Fri, 2010-10-22 at 11:16 +0200, Dave Cotton wrote:
On 22/10/10 11:05, Hans Witvliet wrote:
On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote:
On 21/10/10 22:04, Hans Witvliet wrote:
For suse there is a precompiled version on the OBS (vitsoft)
Package search on the OBS shows
Hi,
I have the same issue. Did you solved it?
On Thu, Jun 18, 2009 at 5:21 PM, Brad Johnson bjohn...@ecessa.com wrote:
We are trying to configure Asterisk (version 1.6.1.0) with some SIP
phones behind a SIP Proxy/NAT device. The phones register properly to
Asterisk, and to get Asterisk to
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 25, 2010 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extension Exists
Hi,
When a VOIP
Bria is a full SIP soft client. It works ok if you have a very good sound
card and good wired headset.
It is not a dialer application in the sense that you would dial your desk
phone using it.
Some of my clients love the Bria and some say the quality is poor. You must
have a computer that can
For those who might be interested...
If possible i rather use maintream prebuild packages.
As from now, they (asterisk180) are available for openSUSE_11.1,
openSUSE_11.2, openSUSE_11.3, SLES10, SLES11, SLES11SP1 via:
http://software.opensuse.org/search?q=asterisk180baseproject=openSUSE%
Hi,
I just downloaded your vdp file and it's working fine on my installation
(Asterisk 1.4).
Can you be more specific on the issue you experienced?
Nile
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Put the outboundproxy=192.0.2.1 under individual sip context not under the
[general], it should work.
CK
On Mon, Oct 25, 2010 at 11:43 PM, sipbeast sipbe...@gmail.com wrote:
Hi,
My asterisk's version is 1.6.0.26. I've couple sip providers and I've
for new SIP provider I need define
On Monday, October 25, 2010 07:30:22 am Stephen Reese wrote:
Does the AGI have to be used? In this example
http://www.davidvossel.com/?p=28 I see mention of a script, but not in
this one:
http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
I believe I missing the connection
Hi,
Is the dev_state can also be used to track a mobile phone's status via SIP? I
tried it on several phones(nokia, samsung) but it returns NOANSWER but i can
hear a beep beep beep sound indicating that it is currently busy.
regards,
RYAN ICASIANO
--
The suggestions did fix the problem. Thank you Shaun and Paul for the help.
Regards,
Jared
On Fri, Oct 15, 2010 at 4:48 PM, Jared Geiger ja...@compuwizz.net wrote:
I haven't heard if this fixed it yet. However I was seeing the echo
cancelers loaded before so I never realized I'd have to do
On 10-10-25 04:21 PM, Dan Journo wrote:
Hi,
When a VOIP user dials an external number, the calls are routed through
our SIP provider.
Is there a simple way to check whether the DDI exists locally before
dialling out to the sip provider?
Something like GotoIfExists(5551...@incoming_calls)
Are these solutions reliable and stable ?.
Have you used these solutions in production ? What about its quality ?
From: Tzafrir Cohen tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Sent: Tue, October 26, 2010 3:12:21 AM
Subject: Re:
I'm planning to use SGM with Asterisk, it is a commercial product.
What is the different between SGM and libs77 and chan_ss7 ? Should I use SGM ?
From: Tzafrir Cohen tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Sent: Tue, October 26, 2010
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