Hello all,
I need that Asterisk Always use Reason in a CANCEL.
How to do?
thank you
*François *
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Hello,
I am using asterisk realtime with a postgresql database on the same server.
In res_pgsql.conf I have specified
[general]
dbhost=localhost
dbport=5432
dbname=asteriskdb
dbuser=psql
dbsock=/tmp/.s.PGSQL.5432
Since both asterisk and db are on same server, I would like asterisk
to connect to
What you mean under this?
You want do something after Dial by the CANCEL reason?
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
When I do a call, and then hangup until ringing, my phone send a CANCEL
with or without a reason header.
I need that asterisk use a reason header in the CANCEL
François
Le 21/05/2010 09:45, Vardan Harutyunyan a écrit :
What you mean under this?
You want do something after Dial by the CANCEL
On Fri, 21 May 2010, Steve Totaro wrote:
On Thu, May 20, 2010 at 7:09 PM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Which bug number have
François BERGANZ schrieb:
Hello all,
I need that Asterisk Always use Reason in a CANCEL.
How to do?
thank you
*François *
hello,
i know that at least version 1.6.1.x send the reason header in a cancel.
best regards
steve smith
--
Für weitere Fragen stehen wir gerne unter v...@sil.at
On 05/21/10 09:07, Leif Madsen wrote:
Danny Nicholas wrote:
If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in
the spirit of your question:
(1) dialplan conversion
(2) loss of functions like Gosub
Can you be more specific about what 1) and 2) mean?
Hello,
Is it possible at the end of a call, to know how many packets were lost
and other QoS informations?
thank you
François
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From alaw to wav, you can use Asterisk's CLI f file
convert /var/lib/asterisk/sounds/soundfile.alaw
/var/lib/asterisk/sounds/soundfile.wav
I want to have my record alaw automically be converted to mp3 (or wav)
right after finishing recording. How can I do it in
Hello list,
I am confronted with the following problem :
making a call only leasts for about 30 seconds, then the call is ended.
The CLI shows :
[May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum
retries exceeded on transmission 954539948-506...@192.168.1.100 for
seqno
Leif - thank you! Will try that.
On Fri, May 21, 2010 at 12:19 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
David Cunningham wrote:
Hello,
We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?
One explanation I've come
Might not be the answer you seek, but you could put a QD AGI in-line to do
the CLI command for you. Something like this:
Exten = s,1,answer
Exten = s,n,record(file.alaw)
Exten = s,n,AGI(alaw2mp3.agi,file.alaw, file.mp3)
Exten = s,n,hangup
Alternatively, you could replace the AGI with a System
Jeff LaCoursiere wrote:
On Thu, 20 May 2010, Gordon Henderson wrote:
On Thu, 20 May 2010, SIP wrote:
Even IF you could get a keyboard with lights you could individually turn
on and off (never seen one),
http://www.artlebedev.com/everything/optimus/
Bit expensive
Hi, List,
I am looking for a cheapest (and therefore most funny) way to attach
GSM card to my asterisk home box.
Needed features:
Calls+SMS in/out
one or two SIM cards (ports)
Should I try looking for a GSM PCI card that is compatible with
linux/asterisk, or GSM USB card, or modern full-blown SIP
Hello!
You can try with a bluetooth usb dongle and a bluetooth phone.
something called chan_mobile. You can get a cheap solution!
have a look.
Regards!
On 21-05-2010 15:19, Motiejus Jakštys wrote:
Hi, List,
I am looking for a cheapest (and therefore most funny) way to attach
GSM card to my
Le 21/05/2010 16:19, Motiejus Jakštys a écrit :
Hi, List,
I am looking for a cheapest (and therefore most funny) way to attach
GSM card to my asterisk home box.
Have a look at chan_mobile (bluetooth connection)
--
Daniel
--
The receptionist at one of our offices called today to stay that there
is an extension on the SPA-932 console that has been solid red for a
few days now. He has restarted the console and we restarted the phone
in question. Still the light stays solid red despite the fact that
there are no calls
2010/5/21 Leif Madsen leif.mad...@asteriskdocs.org
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Which bug number have you reported your issue in?
This is the one
Try core show hints from CLI - since BLF is hint-driven (As best as I know),
you may have a lagging hint.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Anness
Sent: Friday, May 21, 2010 9:58 AM
To:
Olivier wrote:
2010/5/21 Leif Madsen leif.mad...@asteriskdocs.org
mailto:leif.mad...@asteriskdocs.org
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I
met an
issue with BLF-pickup which kept me from going further.
Which bug
I have a Centos 5.4 64 bit installation. I've tried installing asterisk
1.6.2.7 from source, and from RPM, and although overall things work, the
chan_ooh323.so module won't load. Every attempt to load causes Capabilities
failure for OOH323. OOH323 Disabled.
I looked at the source and the
On Fri, May 21, 2010 at 10:11 AM, Danny Nicholas da...@debsinc.com wrote:
Try core show hints from CLI - since BLF is hint-driven (As best as I
know),
you may have a lagging hint.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
2010/5/21 Steve Totaro stot...@first-notification.com
On Thu, May 20, 2010 at 7:09 PM, Leif Madsen leif.mad...@asteriskdocs.org
wrote:
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Which
2010/5/21 Gareth Blades list-aster...@skycomuk.com
Olivier wrote:
2010/5/21 Leif Madsen leif.mad...@asteriskdocs.org
mailto:leif.mad...@asteriskdocs.org
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I
met an
issue with
Assuming that you don't have around the clock phone traffic, it's a good
idea (IMO) to cron a restart when convenient at about midnight local time
each day.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent:
Olivier wrote:
2010/5/21 Gareth Blades list-aster...@skycomuk.com
mailto:list-aster...@skycomuk.com
Olivier wrote:
2010/5/21 Leif Madsen leif.mad...@asteriskdocs.org
mailto:leif.mad...@asteriskdocs.org
mailto:leif.mad...@asteriskdocs.org
For those having the same problem, my solution was to upgrade to the
newest firmware on the Linksys WAG160.
It seemed a NAT-problem because NAT-ting was not correctly handled by
the firmware.
Jonas.
On 05/21/2010 02:41 PM, Jonas Kellens wrote:
Hello list,
I am confronted with the following
Trying to do a FollowMe test. When the extension is dialed, it dials my
cellphone and my cell phone rings. But when I answer my cell phone it's
just silence. When I press '1' on my cell phone, nothing happens.
extensions.conf:
exten = 140,1,FollowMe(mleonetti)
followme.conf
[general]
Hi,
I'm still on 1.4 and am wondering if 1.6 would fix an issue for me.
Specifically, I have been given the impression that, in contrast to 1.4
which always sends packet from the default IP (if the server has multiple
IPs), 1.6 sends packets back from the IP address that was used by the peers.
On Fri, 2010-05-21 at 09:04 +0100, Gordon Henderson wrote:
On Fri, 21 May 2010, Steve Totaro wrote:
On Thu, May 20, 2010 at 7:09 PM, Leif Madsen
I'm still deploying 1.2 - Got one next week with an ISDN-30 connection and
40 seats. It just works and ticks all the boxes I need to tick for
Thanks guys -- it was a call that was hung I did a soft hangup and it
fixed the problem.
Going to set-up that cron job for restarting the service at midnight
as there is no one in the office that late/early.
Thank You,
Steve Anness
On Fri, May 21, 2010 at 10:49 AM, Danny Nicholas
On 05/21/2010 02:48 AM, Deepesh D wrote:
I am using asterisk realtime with a postgresql database on the same server.
In res_pgsql.conf I have specified
[general]
dbhost=localhost
dbport=5432
dbname=asteriskdb
dbuser=psql
dbsock=/tmp/.s.PGSQL.5432
Since both asterisk and db are on
I ended up getting this to work using Endstream for termination with the
same setup. I may try Gafachi again if I can upgrade to 1.6.2 in the near
future as they seems to be recommended for faxing.
On Tue, May 18, 2010 at 9:02 AM, David Backeberg dbackeb...@gmail.comwrote:
On Tue, May 18, 2010
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I had the opportunity to test a Sangoma A200 card and I have some doubts
that I would like to consult:
I tried to detect the card and I had no success using the wctdm module
with DAHDI. I guess this is because electronics is different
On Thu, May 20, 2010 at 11:41 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Have you met
Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I had the opportunity to test a Sangoma A200 card and I have some doubts that
I would like to consult:
I tried to detect the card and I had no success using the wctdm module with
DAHDI.
Did you not bother to
- Daniel Bareiro daniel-lis...@gmx.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
Greetings!
I had the opportunity to test a Sangoma A200 card and I have some
doubts
that I would like to consult:
I tried to detect the card and I had no success using the wctdm
Hi all,
I am looking for a voice recognition technology integrated to
asterisk. Any suggestion about it?
Ango
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New to Asterisk? Join us for a live
I am looking for a voice recognition technology integrated to
asterisk. Any suggestion about it?
I'm using the Vestec product from Digium and having good luck with it.
There's also LumenVox from them as well, but it doesn't support 64-bit
systems, doesn't have good documentation and is more
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