Hi all,
I've been trying to register a SIP user agent to an Asterisk server using
OpenSIPS as SIP router. The functionality is working fine. However,
Asterisk uses the IP address of the OpenSIPS server as the peer IP address.
How can I use the original IP address of the peer without changing the
Thanks
But that's not the problem, I also tried without the quotes and still
the error appears only this time it is like this
app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query
failed. Error: You have an error in your SQL syntax; check the manual
that corresponds to your MySQL
Hi,
We have a problem connecting to a Cisco AS5300 trunk.
We set the sip peer to allow only g729. The call attempt is able to
connect, but when answered, no audio is heard or transmitted.
Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium.
We do not have this problem on
You are hereby encouraged to post your AS5300 IOS config, sip.conf peer
declaration, and packet capture. Those three things would aid greatly in
diagnosis, especially the capture.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general
On Mon, Jan 09, 2012 at 10:09:40AM +0200, Eyal wrote:
On Monday, January 09, 2012 9:56 AM, James Sharp wrote:
On 01/09/2012 02:44 AM, Eyal wrote:
Hi,
I try to create a new table using MYSQL command in asterisk.
This is what i write:
*Query resultid ${connid} CREATE TABLE IF NOT
We have a couple of customers using the Snom MeetingPoint and they are
very happy with them.
On Sun, 2012-01-08 at 12:03 -0500, brya...@zktech.com wrote:
Thank you for your responses. No where did I say I hate polycom phones. I
personally do not like their approach to sip as a company. Their
Hi Alex, here's the config and the sip debug output.
Guide:
xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add
yyy.yy.yy.yy - our asterisk 1.6.2.14 server
sip config:
type=peer
disallow=all
allow=g729
host=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
nat=no
In article acf1979b7d3ca54089c1abda3528b1f901dbc...@media2.media.ltd,
Eyal e...@mcr-m.com wrote:
I try to create a new table using MYSQL command in asterisk.
This is what i write:
Query resultid ${connid} CREATE TABLE IF NOT EXISTS conference_600
(id int(11) NOT NULL auto_increment, channel_id
Hello,
What is the meaning of this DEBUG message?
chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and
clearing call
Thanks,
Elliot
--
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Hello,
http://www.linuxinnovations.com shows the changes and updates between
the various versions of Asterisk.
--Elliot
On Sat, Jan 7, 2012 at 2:10 AM, Joseph syscon...@gmail.com wrote:
On 01/06/12 16:35, Joseph wrote:
On 01/06/12 18:15, Eric Wieling wrote:
Putting in a Wait(n) is only
One thing i have noticed is that your profile-id don't match and therefore you
would get no video. Asterisk is not a problem
On Jan 9, 2012, at 1:20 AM, Durgesh Mishra wrote:
Hi,
I am facing an issue while testing the video mail service of Asterisk. I have
two different setup on one setup
On 02/01/12 04:39 PM, sean darcy wrote:
OK, the book is out of date. Do Not put the name of the local
device/user in the register statement.
Does it work with UDP? If so, then that is a different behaviour. The
book only tested with UDP, not TCP, so if it works with UDP, then it was
working
On 05/01/12 05:24 PM, Kevin P. Fleming wrote:
snip
Although in my personal opinion, it's really
hard to beat the IP5000.
That has been my experience as well.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
--
_
--
O.P. doesn't state his Asterisk version, but in 10.0(beta) I had a similar
problem where sqlite3 couldn't create the new Asterisk DB. From what I read
in the archives, we really could use a guru to thoroughly pound these DB
statements to make them a bit more bullet-proof.
-Original
I am trying to collect information regarding a bug report for Elastix (http://bugs.elastix.org/view.php?id=1146). In this bug, an user has asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an outbound call through an ISDN trunk, by placing
Dial(DAHDI/g0/12345w) in order to send
w is only allowed as part of the dialed TN on FXO and FXS ports.
Dial the TN normally, use the D() option to Dial to send post answer digits.
i.e. Dial(DAHDI/g0/12345,240,D(w))
See core show application dial
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Monday 09 January 2012, Danny Nicholas wrote:
O.P. doesn't state his Asterisk version, but in 10.0(beta) I had a similar
problem where sqlite3 couldn't create the new Asterisk DB. From what I
read in the archives, we really could use a guru to thoroughly pound
these DB statements to make
On Mon, 9 Jan 2012, A J Stiles wrote:
Strict ANSI SQL specifies 'single speech marks' around values, and no
reserved words in field names.
Is this a UK'ism?
I've never seen a 'quotation mark' (single or double) referred to as a
'speech mark.'
--
Thanks in advance,
On 2012-01-09 17:46, Alex Villacís Lasso wrote:
I am trying to collect information regarding a bug report for Elastix
(http://bugs.elastix.org/view.php?id=1146). In this bug, an user has
asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an
outbound call through an ISDN trunk, by placing
Hi,
On a brand new system, I met an issue I've never met before.
My setup is :
debian 6.0.3
asterisk 1.8.8.1
dahdi 2.6.0
libpri 1.4.12
freepbx 2.9.0.4
TE420FB (with hardware EC)
This is the very first time I'm using Freepbx and the whole
configuration was first generated by a make samples
Hello Folks!
I´m trying attended transfer with asterisk 1.6 and see the message
WARNING[] features.c: Failed to play transfer sound! once in a while when
the transfer failes.
Any idea what can be happening?
Thanks a lot!!!
--
_
I am trying to collect information regarding a bug report for
Elastix (http://bugs.elastix.org/view.php?id=1146). In this bug,
an user has asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make
an
outbound call through an ISDN trunk, by placing
Dial(DAHDI/g0/12345w) in
On a brand new system, I met an issue I've never met before.
My setup is :
debian 6.0.3
asterisk 1.8.8.1
dahdi 2.6.0
libpri 1.4.12
freepbx 2.9.0.4
TE420FB (with hardware EC)
This is the very first time I'm using Freepbx and the whole
configuration was first generated by a make
On Mon, Jan 09, 2012 at 07:52:02PM +0100, Olivier wrote:
Hi,
On a brand new system, I met an issue I've never met before.
My setup is :
debian 6.0.3
asterisk 1.8.8.1
dahdi 2.6.0
libpri 1.4.12
freepbx 2.9.0.4
TE420FB (with hardware EC)
This is the very first time I'm using Freepbx
Hi gang,
I'm thrilled to be able to use a better quality sound in
Asterisk 10, but have to change my wav files to sln44 to get the benefit.
Is there some conf setting I'm missing that would let me play a wav at 44
Khz instead of having to do this?
Sox mon-0a.wav h-1a.wav -t
I´m trying attended transfer with asterisk 1.6 and see the message
WARNING[] features.c: Failed to play transfer sound! once in a
while when the transfer failes.
Any idea what can be happening?
Asterisk tried to play the features.conf xfersound configured sound file.
By default this is the
In article alpine.DEB.2.00.1201091001320.16313@localhost.localdomain,
Steve Edwards asterisk@sedwards.com wrote:
On Mon, 9 Jan 2012, A J Stiles wrote:
Strict ANSI SQL specifies 'single speech marks' around values, and no
reserved words in field names.
Is this a UK'ism?
I've never
On Mon, 9 Jan 2012 13:59:07 -0600
Danny Nicholas da...@debsinc.com wrote:
Hi gang,
I'm thrilled to be able to use a better quality sound
in Asterisk 10, but have to change my wav files to sln44 to get the
benefit. Is there some conf setting I'm missing that would let me
What do I need to set to play 16 Khz wav files?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lefteris
Zafiris
Sent: Monday, January 09, 2012 2:32 PM
To: asterisk-users@lists.digium.com
Subject: Re:
El 09/01/12 14:40, Richard Mudgett escribió:
I am trying to collect information regarding a bug report for
Elastix (http://bugs.elastix.org/view.php?id=1146). In this bug,
an user has asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make
an
outbound call through an ISDN trunk, by placing
On Mon, 9 Jan 2012 14:40:47 -0600
Danny Nicholas da...@debsinc.com wrote:
What do I need to set to play 16 Khz wav files?
Rename them to .wav16
Lefteris Zafiris
--
_
-- Bandwidth and Colocation Provided by
On 01/09/2012 02:32 PM, Lefteris Zafiris wrote:
On Mon, 9 Jan 2012 13:59:07 -0600
Danny Nicholasda...@debsinc.com wrote:
Hi gang,
I'm thrilled to be able to use a better quality sound
in Asterisk 10, but have to change my wav files to sln44 to get the
benefit. Is there some
On Mon, Jan 09, 2012 at 01:47:48PM -0600, Shaun Ruffell wrote:
On Mon, Jan 09, 2012 at 07:52:02PM +0100, Olivier wrote:
Hi,
On a brand new system, I met an issue I've never met before.
My setup is :
debian 6.0.3
asterisk 1.8.8.1
dahdi 2.6.0
libpri 1.4.12
freepbx 2.9.0.4
On Mon, Jan 09, 2012 at 10:37:12AM -0600, Danny Nicholas wrote:
O.P. doesn't state his Asterisk version, but in 10.0(beta) I had a similar
problem where sqlite3 couldn't create the new Asterisk DB. From what I read
in the archives, we really could use a guru to thoroughly pound these DB
Luke Hamburg l...@solvent-llc.com writes:
Carlos-
Sorry if this is too much of a digression but this piqued my interest as
I've been pretty happy with Polycom in my limited experience (haven't used
the SPAs much, just Yealink Polycom, and an occasional Snom here and
there). If the config
On Sun, Jan 08, 2012 at 02:30:03PM +0200, Tzafrir Cohen wrote:
Is there any way to tell Asterisk to flush that cache? Is this
considered a bug?
It's a feature. To enable such polls, set:
pollmailboxes = yes
;pollfrequency = NN ; seconds
Thanks to mjordan on IRC.
--
You got me. At first the polycom world was hard to get into. But with a little
effort to understand the configs and the joys of central provisioning the
Polycom are my go to endpoint. Couple the endless configurablity with Polycom
quilaty and I have many happy clients.
As an aside is that what
On Sun, Jan 8, 2012 at 12:03 PM, brya...@zktech.com wrote:
Thank you for your responses. No where did I say I hate polycom phones. I
personally do not like their approach to sip as a company. Their audio
quality is top notch but for me the rest leaves me wanting. Has anyone used
the newer
On Sun, Jan 8, 2012 at 3:38 PM, Carlos Alvarez car...@televolve.com wrote:
On Sun, Jan 8, 2012 at 9:02 AM, C F shma...@gmail.com wrote:
I find that the bottom line of all polycom haters is ones inability of
comprehending the config files and not in its quality.
We have no problem with
Exactly which IE message are you trying to push manually? you
shouldn't have to do that, it should be done in the configs for you.
On Mon, Jan 9, 2012 at 1:45 PM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
On 2012-01-09 17:46, Alex Villacís Lasso wrote:
I am trying to collect
Here's the cisco AS5300 settings from our provider
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g723r53
codec preference 4 g723r63
codec preference 5 g723ar53
codec preference 6 g723ar63
On Mon, Jan 9, 2012 at 5:18 PM, Roi Stork roi.st...@gmail.com wrote:
Hi Alex,
The problem has been fixed.
We are able to hear audio in our calls after adding these lines in the
AS5300 config:
sip-ua
g729-annexb override
There's an issue regarding codec matching in IOS versions 12.3(18) or higher:
https://supportforums.cisco.com/docs/DOC-3186
On Tue, Jan 10, 2012 at
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