On Sun, 6 Feb 2011 16:27:33 -0600, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Have you tried playing with the length of the wait? Even if you technically
need 10 seconds, you could try a lower amount to see if the other priorities
in that context execute...
Lowering it to 5 seconds makes
On Mon, Feb 7, 2011 at 2:22 AM, Gilles codecompl...@free.fr wrote:
Lowering it to 5 seconds makes no difference. I also tried adding a
Hangup before Wait but then the script ends before Wait.
That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
DAHDI/Zap setup... Barring
On Mon, 7 Feb 2011 02:59:09 -0600, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
DAHDI/Zap setup... Barring something in your configuration that I don't know
about, there's no reason that the system should just hang up the
... but Asterisk does nothing, altough show modules says that
pbx_spool.so is loaded. Weird :-/
FWIW, Asterisk runs as root, and root owns callfile.call.
Maybe it's the uClinux or the Asterisk I'm using that's configured in
such a way that callfiles don't work as planned.
Apparently,
On 02/06/2011 05:05 PM, Sherwood McGowan wrote:
AAhem.
https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT
Granted, it's in 1.8, but it's in the documentation ;-)
Cheers
That seems to do exactly what the Lobstertech code does. What do people
use this for? The
Hi,
I am using Trixbox 2.6.2.3, ISO install
I am getting the below error in `/var/log/asterisk/full`
Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Is there anyway to figure out which extension is causing this error ?
Thank you.
Best regards,
Sanjay
--
On Mon, Feb 7, 2011 at 7:42 AM, RSCL Mumbai rscl.mum...@gmail.com wrote:
Hi,
I am using Trixbox 2.6.2.3, ISO install
I am getting the below error in `/var/log/asterisk/full`
Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Is there anyway to figure out which
On Mon, 7 Feb 2011 04:06:52 -0600, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
ok, first of all, it can take a little while for those spooled callfiles to
be executed in Asterisk...
Thanks for your help. The same callfile works fine in Ubuntu, but not
at that appliance. Since I can dial
Dear list,
I want to avoid sending calls to a queue when it is full. From the fact that
'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like
to know if there's a way to do it. Setting the Queue() timeout to a little
value is not the most suitable option.
I'm using asterisk
On Mon, Feb 7, 2011 at 8:39 AM, Steve Underwood ste...@coppice.org wrote:
On 02/06/2011 05:05 PM, Sherwood McGowan wrote:
AAhem.
https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT
Granted, it's in 1.8, but it's in the documentation ;-)
Cheers
That seems to do
This might be what you're looking for
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action
+QueueStatus
Ish
On Mon, 2011-02-07 at 10:44 -0500, Daniel - Asterisk wrote:
Hi Danny,
Could you please let me know what function do I use to get if the
queue is full?
Elder
On
I`m currently having aleatory call drops through a PRI and so I want to
upgrade libpri.
In case I have errors I will want to downgrade libpri, that`s the reason why
I asked.
Asterisk version: 1.6.2.13
Current Libpri: libpri-1.4.10.2
Libpri to be upgraded to: libpri-1.4.11.5
Any help
Bruce B wrote:
Oh, no problem. I do understand that there are a bunch of you guys who
would like to jump on a topic with smart-/aleck remarks and /act as
moderators of the list. This is nothing new to me and I understand
that you totally defy the whole purpose of the mailing list. Keeping
Hi,
The following IAX config (slightly edited) causes an issue for me in
version 1.6.2.16.1, where my CDR data is unreliable.
[user1]
type=friend
auth=md5
accountcode=user1
notransfer=yes
context=context1
host=10.0.0.250
username=user1
secret=secret1
disallow=all
allow=alaw
[user2]
type=friend
On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards
asterisk@sedwards.com wrote:
sudo /usr/sbin/asterisk -d -d -d -n -v -v -v
Oops. A '-c' should be in there :)
Thanks Steve for the help.
I launched * with asterisk -d -d -d -n -v -v -v -c, and ran module
show to check that
On Saturday 05 Feb 2011, Timothy Smith wrote:
On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
Can you listen to an mp3 file through the Asterisk server's own sound
card (if it has one; if not, use the -w option to write to a .wav file,
and test that by
On Mon, Feb 07, 2011 at 02:13:32PM -0200, Agustina Berretta wrote:
I`m currently having aleatory call drops through a PRI and so I want to
upgrade libpri.
In case I have errors I will want to downgrade libpri, that`s the reason why
I asked.
Upgrading? Downgrading?
Asterisk version:
On Mon, Feb 7, 2011 at 10:46 AM, Gilles codecompl...@free.fr wrote:
=
#callfileSIP.call
Channel: SIP/xlite
Context: callback-dialtone-auth
Extension: s
Priority: 1
MaxRetries: 2
RetryTime: 60
WaitTime: 30
=
Just a thought...
Did you originally generate this
Real quick, please respond to my question about where the callfile ends up
after a few minutes, as well as the modification time and the permissions on
the file ;-) These are good bits to know
On Mon, Feb 7, 2011 at 10:46 AM, Gilles codecompl...@free.fr wrote:
On Mon, 7 Feb 2011 07:57:07 -0800
Bruce, it's all good my man
Hey Novack...while I'm sure that both of you are quite pleased with your own
witty smart-aleck remarks, let me first say, in the past I've sat back (for
quite some time, seeing as how I've been on this list for close to what 5 or
6 years under varied emails throughout
oh and didn't you guys already have your little histrionics sessin about
trimming the goddamned emails, mailing list etiquette about top posting
versus bottom, etc../..
My complaint is not something as trivial as where one should reply in a
mailing list email, or if one should trim emails every
Hi List,
Quick question:
I am using asterisk 1.8.1 - how do I detect whether a native (remote) bridge is
being used between 2 SIP peers?
It is not obvious to me from the console logs.
Thanks,
Ondrej
--
_
-- Bandwidth and
On 02/07/2011 11:46 AM, Gilles wrote:
snip
Asterisk runs as root, and owns this file as well.
Have you tried setting the permissions of this file to world readable,
to ensure that any user can read it and eliminate potential permissions
problems?
Worth a shot. While you're at it, output
Actually mate, I'm about to start a run of testing on a project that
actually applies...if I figure it out shortly, I'll respond :D
On Mon, Feb 7, 2011 at 11:49 AM, Ondrej Valousek webs...@s3group.cz wrote:
Hi List,
Quick question:
I am using asterisk 1.8.1 - how do I detect whether a
Hi List,
I am using asterisk 1.8.1. and I want to avoid transcoding when 2 SIP peers
calling each other:
A (g722, alaw) calls B (alaw,ulaw) via asterisk.
My setup:
allow=g722,alaw
preferred_codec_only=no
Note that when B calls A, codec alaw is used on both ends, fine, but it does not seem
On Mon, Feb 7, 2011 at 12:40 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
oh and didn't you guys already have your little histrionics sessin about
trimming the goddamned emails, mailing list etiquette about top posting
versus bottom, etc../..
My complaint is not something as
Asterisk runs as root but what about the bash script or the php file that
creates the file? Maybe comment the mv command and check the file
permissions by *ls -la call-filename.call* to be sure.
*chown root.root call-filename* (if root is really the user running
Asterisk) and then the mv command
*** ever so politely snipping ***
If you are sure that permissions are not the problem and you have archive
set to yes then you can browse the */var/spoo/asterisk/outgoing_done*folder
to see if the call file is transferred there or not. The file should
contain some info to help you and it's
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood
McGowan
Sent: Monday, February 07, 2011 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback through
Does anybody know of a Similar list for SwitchVoX?
And would like to post to proper list if one is available.
I had posted on digium forum, but have not received any responses yet.
http://forums.digium.com/viewtopic.php?f=38
Try these -
SwitchVox SMB and SOHO
http://forums.digium.com/viewforum.php?f=38
http://forums.digium.com/viewforum.php?f=38sid=e78c5fda089b88d8e1617d0c548
d8f8d sid=e78c5fda089b88d8e1617d0c548d8f8d
SwitchVox Free Editiion Support
http://forums.digium.com/viewforum.php?f=19
Bruce,
All in all, I don't think it's that hostile, it just goes through
cycles...maybe a good number of us may indeed have estrogen issues and it's
the moon, who knows ;-) LOL
Cheers (and I always mean it, seriously :D )
Sherwood McGowan
Yes, THAT Mick
--
Hi everybody,
I have two toll free numbers pointed to my asterisk server. My toll free
number provider gave me two 7 digit dnis numbers. Both numbers land in the
extensions.
How to make the softphone (xlite) know that the call has landed through
which number? I think the differentiating stuff is
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohan Shahi
Sent: Monday, February 07, 2011 12:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] multiple inbound calls from same sip trunk
Hi everybody,
I
In my (1.4.X) experience, the file just stays in
/var/spool/asterisk/outgoing and gets “little tags” added until you get the
problem resolved or delete the file.
That is absolutely true if the file is not processed. I guess he can again
do a ls -la in that folder to check permissions for
Hi Group, Do you think this has been fixed or it's still not supported with
standalone NOTIFY?
Your help will be highly appreciated!
Regards,
Felton
- Original Message
From: Feng Xu felto...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Fri, 4 February, 2011 1:43:39 PM
Hi,
Here is a solution to your problem.
By default asterisk send all OPTION messages to default context in dialplan
regardless on peers's context. You will get 200 OK reply to your option packet
if you add following lines to dialplan,
[default]
exten = _X.,1,NoOp()
exten = _X.,n,Hangup()
Hi,
If you need full control on both legs of call you can redirect Leg-1 to your
dialplan as [mailto:Local/your-extension@your-context/n] Channel:
Local/your-extension@your-context/n and from there you control the Leg-1 using
dial-plan or AGI as you like while Leg is normally comes to
Hi,
If you will send call without answering on asterisk and have directrtpsetup=yes
in sip.conf codec negociation will always be between UAs so any matched codec
will work fine. If you are answering call on asterisk then dialing it out to
next UA then you need to add canreinvite=yes for both
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