did you try rebooting after installing 11.9?
-Original Message-
From: Administrator TOOTAI ad...@tootai.net
Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 30 Apr 2014 15:13:59
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
You could have the call immediately return to the transferer
-Original Message-
From: John Kiniston johnkinis...@gmail.com
Sender: asterisk-users-bounces@lists.digium.comDate: Thu, 6 Feb 2014 17:14:02
To: Asterisk Users Mailing List - Non-Commercial
Try setting canreinvite yes on that trunk it worked on trunks I had
Some providers send a reinvite after 15 min and if Asterisk doesn't respond
then it disconnects the call something like that
-Original Message-
From: Jonas Kellens jonas.kell...@telenet.be
Sender:
Are you using a mp3 file?
I have noticed that using control playback with a mp3 file I cannot use the
keypad to control the playback
-Original Message-
From: Salaheddine Elharit salah.elharit...@gmail.com
Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013 08:05:16
Thanks for replying (I only asked on this list)
Whatever function you add to that file becomes a function and that was a odbc
function I added
Anyhow after a restart of asterisk it started working ok
It worked like a charm (I had more than 5 inserts to a database within a
few hours)
In asterisk.conf you need to enable running of eternal scripts
-Original Message-
From: Asmaa Ahmed asabatg...@hotmail.com
Sender: asterisk-users-bounces@lists.digium.comDate: Sun, 29 Sep 2013 16:48:32
To: Asterisk Users Mailing List - Non-Commercial
In asterisk.conf you need to enable running of external scripts
-Original Message-
From: Asmaa Ahmed asabatg...@hotmail.com
Sender: asterisk-users-bounces@lists.digium.comDate: Sun, 29 Sep 2013 16:48:32
To: Asterisk Users Mailing List - Non-Commercial
Some providers send a reinvite after 15 min and if asterisk doesn't respond
will disconnect the call
Maybe playaround with canreinvite
--Original Message--
From: Jeremy Kister
Sender: asterisk-users-boun...@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Did you the a2billing settings for a music on hold setting
I remember seeing some setting
-Original Message-
From: Nick Cameo sym...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 10 Sep 2013 12:46:54
To: Asterisk Users Mailing List - Non-Commercial
How about sending the whole path to mutt in the system call
System(/usr/sbin/mutt) where ever it is
-Original Message-
From: Ishfaq Malik i...@pack-net.co.uk
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 21 Jun 2013 08:49:30
To: Asterisk Users Mailing List - Non-Commercial
I think facebook uses xmpp so you could use asterisk jabber or so
Don't know about the rest
-Original Message-
From: bilal ghayyad bilmar...@yahoo.com
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Date: Wed, 17 Apr 2013 14:41:53
To: asterisk-users@lists.digium.com
Reply-To: Asterisk
Try canreinvite=yes in sip trunk
-Original Message-
From: Florian Wolters flor...@florian-wolters.de
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 21 Mar 2013 08:31:54
To: asterisk-users@lists.digium.com
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Check out connectedline()
-Original Message-
From: Rusty Newton rnew...@digium.com
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Date: Tue, 19 Feb 2013 09:58:30
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing
Look at asterisk 11
A option was added to play announcements between music Files and so forth
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Can't we. Do this?
exten = 520xx/0666XX,1,hangup
-Original Message-
From: Salaheddine Elharit salah.elharit...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 14 Jan 2013 16:51:11
To: Asterisk Users Mailing List - Non-Commercial
--Original Message--
From: Eric Wieling
To: ישראל גוטליב
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] block one number in incoming calls
Sent: Jan 14, 2013 6:58 PM
No. However you can do this: exten = _520xx/_0666XX,1,hangup
Did you set externip and localnet in your sip conf ?
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Just my pitch in to post
From a blackberry you can only top post there is no way of bottom posting
So if I would have to wait to get to a computer to bottom post I would just
never answer
-Original Message-
From: Carlos Alvarez car...@televolve.com
Sender:
They mentioned some time back about redoing the design site so that might be
the reason
-Original Message-
From: Justin Killen jkil...@allamericanasphalt.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 17 Dec 2012 14:54:57
To: Asterisk Users Mailing List - Non-Commercial
Hi,
If were on this subject I'll throw in my question
Does named acl lists in asterisk 11 help for this or only for registrations?
Thanks,
-Original Message-
From: Joshua Colp jc...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 26 Nov 2012 10:28:05
To: Asterisk
Thought so but hoped other wise
Thanks
--Original Message--
From: Joshua Colp
To: ? ??
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No matching peer for 'callerID' from
'85.xx.xx.2:5060'
Sent: Nov 26, 2012 4:40 PM
Check the notifyringing option in sip.conf
-Original Message-
From: Chris Owen ow...@hubris.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 22 Oct 2012 15:17:27
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk
Did you try restarting asterisk not only a reload
Also I found a few broken stuff in queues like the rules (yes its on the
tracker) maybe this is also
-Original Message-
From: Mitch Claborn mitch...@claborn.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 27 Sep 2012
Hi,
I have 10 agents who are pstn lines in queue and would like that when they
answer the rtp should go directly
Is it at all possible in queues?
If yes what could be bothering it from happening?
Thanks,
Israel
--
_
--
She's talking about asterisk 11 not asterisk 1.8.11
-Original Message-
From: Phil Frost p...@macprofessionals.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 21 Aug 2012 15:19:31
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
add a /n at the end of the local channel
-Original Message-
From: Rodrigo Lang rodrigoferreiral...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 1 Aug 2012 15:53:44
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To:
He's probably using softphones
-Original Message-
From: Kevin P. Fleming kpflem...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 06 Jul 2012 13:32:20
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Did you install the addons
Yum install asterisk18-addons-mysql
-Original Message-
From: Duncan Turnbull dun...@e-simple.co.nz
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 17 Jun 2012 08:30:00
To: Asterisk Users Mailing List - Non-Commercial
Hi all
I'm trying to get asterisk 10 spandsp get faxes from 012 in israel (they use
broadsoft switches) using T.38 more reliable and would like to know if anyone
knows of any changes I could make or ask them to make.
As it stands now I get much more reliability receiving faxes with iaxmodem
Of course you are disabling the video maybe also include the video protocols in
the sip_codec
-Original Message-
From: Tarek Sawah tareksa...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Sat, 19 May 2012 17:33:57
To: Asterisk Usersasterisk-users@lists.digium.com
Broadvoice has a lot of problems for the last 2 months
-Original Message-
From: Ing. CIP Alejandro Celi Mariategui a...@linux.org.pe
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 04 May 2012 02:11:11
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List
The asterisk side has to have the router ports 5060 and 1-2 forwarded
to asterisk these are the standard ports but you could cut way down on the rtp
ports in rtp.conf then you have to tell asterisk what's the external ip of
your nat and most of the times this should work today no
Well you have to tell asterisk what's the external ip of the nat else its never
gone work
Look at externip and localnet
-Original Message-
From: Carlos Alvarez car...@televolve.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 26 Apr 2012 14:15:39
To: Asterisk Users Mailing
יעע
-Original Message-
From: Vieri rentor...@yahoo.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 17 Apr 2012 23:27:10
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject:
Do you have r in your dial string?
If yes remove that
-Original Message-
From: Leandro Dardini ldard...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 25 Mar 2012 11:35:45
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
There is a bug in up to version 1.8.9 with external moh sources and dahdi timers
-Original Message-
From: Stephen Brown stephen.brow...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 21 Feb 2012 15:34:19
To: Asterisk Users Mailing List - Non-Commercial
You could preload the res_moh (don't remember the full name) but that will only
help until the next reload which is the next time you'll click the orange bar
Or use a different timer which could get you into other problems
Maybe some else has a other idea
-Original Message-
From:
Hi,
Does anyone how I could extract redirected number from a sip packet
I have redirected a cell to a second cell which also rings a sip trunks and
wish to route the call per rdnis
The rdnis variable brings the first redirect (divert) which is the second cell
but the first number also appears
On the snom too
Create a conferance and then press the transfer button. That will join the
parties and release the receptionist
-Original Message-
From: Andres and...@telesip.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 14 Feb 2012 17:10:38
To: Asterisk Users Mailing
Your running into a bug and the only way to solve it is to report it and debug
it and hope for a fix
There is no way someone can help without it being debugged and knowing what's
causing it to lockup
The only key to unlcock it when it gets locked is by restarting asterisk
Regards
M…
-Original Message-
From: Kingsley Tart kings...@skymarket.co.uk
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 01 Feb 2012 10:34:07
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
The new snom 7 series and maybe the 8 series have Gig ethernet
-Original Message-
From: Vieri rentor...@yahoo.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 13 Jan 2012 04:45:12
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
Does anyone know what languages are supported?
-Original Message-
From: Bruce B bruceb...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 4 Jan 2012 13:25:18
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk
Rename the wav to ulaw
Miss_audio.ulaw
-Original Message-
From: shalu dhamija shalu.dham...@rancoretech.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 26 Dec 2011 10:48:36
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
Well freepbx has that in the gui you should read the tool tips
Read the trunk limit tooltip
-Original Message-
From: Steve Edwards asterisk@sedwards.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 20 Dec 2011 12:16:48
To: Asterisk Users Mailing List - Non-Commercial
The variable for outbound is (SIP_CODEC_OUTBOUND=g722)
But I think asterisk will try to transcode then because the preferred codec on
the phone is ulaw or so
-Original Message-
From: Danny Nicholas da...@debsinc.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 15 Nov 2011
There is a bug which blocks call progress message 8 which was fixed but I
don't remember in which version
Try upgrading to latest 1.6 version
-Original Message-
From: cb c...@mythtech.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 8 Nov 2011 09:51:40
To: Asterisk Users
A telco could either give you a analog line like the old phone line which you
have at home with 1 number and 1 line or a T1 which comes from the telcos
office to yours and plugs directly into a digital gateway with 23 lines and
lots of numbers. and no need at all for analog gateways on the way
The mp124 is a analog gateway and doesn't support t1's I think
A T1 is a digital line which has 24 channels per port which means 24 calls
concurrently if you want more channels you need more ports
DID's are incoming numbers the telco sends down your trunk(port) you could have
thousands of
Thanks bryant ur right 23 channels I'm used to E1's where you a get 30 channels
a even number
-Original Message-
From: Bryant Zimmerman brya...@zktech.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 3 Nov 2011 22:32:41
To: Asterisk Users Mailing List - Non-Commercial
Where do you see that ?
In the log you sent its setting the callerid and then dialing
-Original Message-
From: motty.cruz motty.c...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 27 Oct 2011 16:02:46
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users
Is the ivr using early media?
-Original Message-
From: Anton Kvashenkin anton.juga...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 17 Oct 2011 12:08:51
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk
Search for dialog-info pickup
-Original Message-
From: Marek Cervenka cerv...@fpf.slu.cz
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 07 Oct 2011 09:47:45
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users
It doesn't work at all with the dahdi timers
The reason it works it works till the first reload is because you are
preloading it before dahdi so it starts and uses the pthread timer later when
you reload it starts using the dahdi timer and there it goes
-Original Message-
From: Luke
I'm just throwing in my 2c (I don't have polycom)
Are your phones auto provisioned then maybe the provisioning server is sending
a reboot for some reason or maybe something on the server is sending a sip
notify of reboot
-Original Message-
From: Gord Urquhart gord...@gmail.com
Sender:
You should change in dahdi conf the amount of time (rings) it should wait
before answering
The dialplan doesn't handle that
-Original Message-
From: Ruben Rögels ruben.roeg...@jumping-frog.org
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 05 Aug 2011 12:36:46
To: Asterisk
Well even in my example there is a mistake in the second line change the 1 to a
2
exten =_.,1,Set(VOLUME(TX)=10)
exten =_.,2,Set(VOLUME(RX)=10)
-Original Message-
From: Zeeshan Ali Shah zees...@infoshield.info
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 5 Aug 2011
If we are talking about adding stuff to the repo I would vote for jabber and
gtalk also fax (spandsp)
-Original Message-
From: Bruce B bruceb...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 2 Aug 2011 13:36:31
To: Asterisk Users Mailing List - Non-Commercial
You could force g711 inbound by using
Set(SIP_CODEC=ulaw)
-Original Message-
From: Kevin P. Fleming kpflem...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 27 Jun 2011 14:08:00
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
You could use the c option in the dial command which sends a call answered
elsewhere reason to the phone and then the phone won't record it in the missed
list (I know it works on the snom I didn't check it on the yealink )
But you'll have to send that only with the dial command which you don't
-Original Message-
From: Steve Totaro stot...@asteriskhelpdesk.com
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Date: Fri, 10 Jun 2011 06:30:53
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
Remove the trailing period after the 5 if that's your whole number
-Original Message-
From: Satish Patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 30 May 2011 14:09:56
To: Asterisk Users Mailing List - Non-Commercial
Run Service dahdi start
-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 16 May 2011 18:41:01
To: asterisk-usersasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Sorry for top post I'm responding from my blackberry
I haven't tried with timerfd but with timer pthread 1.8 is very unstable
I think I have seen a post to the list from kevin fleming that the same is for
timerfd that there is a nasty bug which they haven't found the reason for yet
Hi,
I'm trying to add modules compiled from source into a rpm install of asterisk
(from digium) on centos and asterisk complains that its not compiled with same
options so it won't load it
I know I could install the entire thing from source but for other reasons I
would like to keep the main
https://issues.asterisk.org/view.php?id=18868
-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 8 May 2011 11:43:41
To: asterisk-usersasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
Look at function CURL
-Original Message-
From: Daniel Isenmann daniel.isenm...@seetec.de
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 6 May 2011 13:04:09
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
That should be CUT all caps I think
-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 7 Apr 2011 20:45:21
To: asterisk-usersasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
Change Wait,2 to wait(2)
-Original Message-
From: bilal ghayyad bilmar...@yahoo.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 5 Apr 2011 01:31:11
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Also change DeadAGI,a2billing.php to AGI(a2billing.php)
-Original Message-
From: bilal ghayyad bilmar...@yahoo.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 5 Apr 2011 01:31:11
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
Ok thanks I found the problem
The spa8000 has some bugs with t38 which are fixed in the spa2102 but not in
the 8000
1. If the adapter starts with g711 It doesn't switch to t38
2. (This my problem) when it does go to t38 and the itsp asks for it to
fallback to 9600 it doesn't fallback so
So make a whitelist
What I do is create a outbound route with the allowed cid and then have another
route which goes to a not allowed recording which catches all other caller Id's
-Original Message-
From: Peter den Hartog peterdenhar...@gmail.com
Sender:
Shouldn't that be
Exten = 1104, 1, Goto(smvoice-mediaport-public-address,s,1)
-Original Message-
From: Rizwan Hisham rizwanhas...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 15 Mar 2011 19:03:33
To: Asterisk Users Mailing List - Non-Commercial
I think he meant the opposite he is sending calls to a sip trunk and would like
to know when to failover and send calls to a different sip trunk
I haven't really looked at this but maybe check the header of the packet for
which response your getting
Also are you sure you are getting the
As far I know asterisk doesn't handle the publish sip dialog so it just keeps
it hanging around in 1.8.X (in previous versions it didn't)
I turned off all publish dialogs in the snom phones I have and that got rid of
that
It doesn't really have any impact on the system as far as I have seen
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