Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
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 asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ? Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! Le 30/04/2014 13:00, Administrator TOOTAI a écrit : Le 30/04/2014 12:39, Administrator TOOTAI a écrit : Le 30/04/2014 12:15, Administrator TOOTAI a écrit : Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards We checked on a customer installation made one week ago: they have the same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server. And finally the explanation: if you modify sip.conf file, the reload is taken in account, all is good. But if the sip.conf contains includes and you modify one of those includes *without modifying* sip.conf, no reload. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rejecting a call as if the extension does not exist.
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Rejecting a call as if the extension does not exist. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call duration limit ? Calls end after 15 minutes...
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: asterisk-users-bounces@lists.digium.comDate: Wed, 08 Jan 2014 16:07:22 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Call duration limit ? Calls end after 15 minutes... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with speech in IVR
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 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] issue with speech in IVR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error cant write to function ODBC_DEVICES
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) -Original Message- From: Rusty Newton rnew...@digium.com Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 23 Oct 2013 12:15:50 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] error cant write to function ODBC_DEVICES On Sun, Oct 20, 2013 at 6:12 AM, Israel Gottlieb isr...@gmail.com wrote: snip but from the dialplan gives me a error cant write to function ODBC_DEVICES happy to hear any ideas I don't use func_odbc on a regular basis, but from looking at the sample file and looking at the functions provided within Asterisk. The ODBC_DEVICES function does not exist. The three functions available for configuration within func_odbc.conf appear to be ODBC_SQL,ODBC_ANTIGF,ODBC_PRESENCE Also the only examples of the string ODBC_DEVICES out on the web according to Google show up at the various forums you have asked about it. :) So.. can't write to function is definitely expected behavior. Hope that helps! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem to get MWI working
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problem to get MWI working -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem to get MWI working
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problem to get MWI working -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes
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 ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes Sent: Sep 10, 2013 10:23 PM On 9/10/2013 7:05 AM, Administrator TOOTAI wrote: I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk Just for kicks, I would disable session-timers to see if the problem goes away. in the general section and/or each peer in sip.conf: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No remote address on RTP instance - On Ringing
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No remote address on RTP instance - On Ringing I have no idea where the `m` is coming from. I even looked into the A2Billing script. Still digging -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing a fax with mutt does not succeed
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
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mailing a fax with mutt does not succeed
On Wed, 2013-06-19 at 13:03 -0500, Daniel - Asterisk wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the
Linux shel it works, but when trying with Asterisk's System command it
doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com
-a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by
root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
--
I' though I sent this once already but here goes again...
Have you tried changing to asterisk user and then do the same command?
i.e.
sudo su asterisk
You may find that the asterisk user doesn't have a full shell
environment and you will ahve to set one using
chsh
Regards
Ish
--
Ishfaq Malik i...@pack-net.co.uk
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552
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Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
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 Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 17 Apr 2013 14:41:53 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
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 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Pickup how to display CND of incoming number
Check out connectedline() -Original Message- From: Rusty Newton rnew...@digium.com Sender: asterisk-users-boun...@lists.digium.com 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 List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call Pickup how to display CND of incoming number - Original Message - From: David C Klaverstyn david.klavers...@intergraph.com Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I’m not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. You are placing a call *to* Asterisk, therefore the handset, like most will show the number you dialed. I don't know how you would get the CallerID to update during a connected SIP session. I'm no SIP expert, but Googling around - I don't think it's possible, at least easily... http://forums.asterisk.org/viewtopic.php?f=1t=71351p=136777 http://forums.digium.com/viewtopic.php?p=152753 -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MoH with message on intervals
Look at asterisk 11 A option was added to play announcements between music Files and so forth -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] block one number in incoming calls
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] block one number in incoming calls -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] block one number in incoming calls
--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 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of isr...@gmail.com Sent: Monday, January 14, 2013 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] block one number in incoming calls 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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] block one number in incoming calls -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users /div-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
Did you set externip and localnet in your sip conf ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
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: asterisk-users-boun...@lists.digium.com Date: Sun, 30 Dec 2012 19:58:20 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX website
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] FreePBX website -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'
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 Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060' Administrator TOOTAI wrote: Hi list, Hola, I face the following problem on incoming calls from my provider which uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are not sended to the context set in provider sip.conf definition, but are going to the default context setted in [general]. Provider uses few IP's for incoming calls which are not the one used for register. You will need to create separate SIP peers that match on each IP address and direct them accordingly to the correct context. A secondary option is to enable anonymous guest support, but I would not recommend that as it can pose a security risk. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'
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 isr...@gmail.com wrote: 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? ACLs don't control SIP peer matching, so no. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Presence for Offhook/Onhook Only
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 Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Call Presence for Offhook/Onhook Only -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QUEUEHOLDTIME always zero
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 09:20:08
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] QUEUEHOLDTIME always zero
I am also writing an AMI application that will allow management to see
the queue status from an external program and saw the same issues with
the AMI data. Using AMI I am able to get what I need from the
individual records for each queued call.
Mitch
On 09/26/2012 04:09 PM, Mitch Claborn wrote:
Asterisk 1.8.10.1~dfsg-1ubuntu1
Trying to build a simple announcement of the queue status. QUEUEHOLDTIME
is always zero. What am I doing wrong?
queues.conf
[general]
autofill=yes
shared_lastcall=yes
[StandardQueue](!)
musicclass=default
strategy=rrmemory
joinempty=no
leavewhenempty=yes
ringinuse=no
announce-frequency = 30
min-announce-frequency = 15
announce-holdtime = yes|no|once
announce-position = limit
announce-position-limit = 5
announce-round-seconds = 10
setinterfacevar = yes
setqueueentryvar = yes
setqueuevar = yes
[sales](StandardQueue) ; create the sales queue using the parameters in
the StandardQueue template
extensions.conf
exten = 812,1,NoOp(queue status)
same =n,Set(LOGGEDIN=${QUEUE_MEMBER(sales,logged)})
same =n,Set(READY=${QUEUE_MEMBER(sales,ready)})
same =n,Set(WAITING=${QUEUE_WAITING_COUNT(sales)})
same =n,Set(STUFF=${QUEUE_VARIABLES(sales)})
same =n,Verbose(waiting: ${WAITING} calls in queue: ${QUEUECALLS}
avg hold: ${QUEUEHOLDTIME} logged in: ${LOGGEDIN} ready: ${READY})
Regardless of how long a caller has been waiting in the queue, the
output is:
-- Executing [812@LocalSets:1] NoOp(SIP/08000F3BE07C-0048,
queue status) in new stack
-- Executing [812@LocalSets:2] Set(SIP/08000F3BE07C-0048,
LOGGEDIN=1) in new stack
-- Executing [812@LocalSets:3] Set(SIP/08000F3BE07C-0048,
READY=1) in new stack
-- Executing [812@LocalSets:4] Set(SIP/08000F3BE07C-0048,
WAITING=1) in new stack
-- Executing [812@LocalSets:5] Set(SIP/08000F3BE07C-0048,
STUFF=0) in new stack
-- Executing [812@LocalSets:6] Verbose(SIP/08000F3BE07C-0048,
waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1) in new
stack
waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1
--
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[asterisk-users] Queue and reinvite
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 - BLF on Custom devices
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 Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 11 - BLF on Custom devices -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with callfile and CDR
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: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem with callfile and CDR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk trying to call a queue with no members
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 asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk trying to call a queue with no members On 07/06/2012 12:36 PM, Antonio Modesto wrote: I don't want the users to manually login in the queue, I want they join the queue when they turn on their phone. I thought that this was the right way of doing it, how can I do it? That's a reasonable way to do it if you like, although it's pretty uncommon for users to have 'turn on' their phones. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10 T.38 and 012 in Israel
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 with no T.38 which is funny than when having asterisk receive using T.38 and if I try using gateway mode to iaxmodem then its even worse. I'm trying to test how much reliability will I loose when moving from a pri which is excellent but expensive to voip and T. 38. How much tradeoff Thanks in advance, Israel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SET SIP_CODEC and Video issues
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 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SET SIP_CODEC and Video issues -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadvoice Got SIP response 503 Service Unavailable
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 - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Broadvoice Got SIP response 503 Service Unavailable Hi, I'm running Asterisk 1.8.11.1 @office. The Broadvoice service work fine with all 1.6 version and early 1.8 behind a NAT but about 2 months ago stop working. No made changes in the firewall NAT rules. Right now I'm @home via my Xlite softphone working fine without problems Any suggestions or thoughts? Alex Celi This is the info central*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status 488/488181.64.96.122D 11037OK (182 ms) sip.broadvoice.com/305422 206.15.148.221 5060 OK (131 ms) sip.conf externip=190.12.68.20 localnet=192.168.20.0/255.255.255.0 localnet=192.168.10.0/255.255.255.0 nat=comedia pedantic=no register = 3054221...@sip.broadvoice.com:XX:3054221...@sip.broadvoice.com [sip.broadvoice.com] type=friend host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3054221494 defaultuser=3054221494 authname=3054221494 secret=X context=entrantes dtmfmode=inband dtmf=inband nat=comedia directmedia=no qualify=yes callgroup=1 pickupgroup=1 disallow=all allow=ulaw allow=alaw I turned on sip debug. This is what I received 181.64.96.122: Is my home IP 190.12.68.20 or central.cipher.pe: is office IP 206.15.148.221: Broadvoice Server --- SIP read from UDP:181.64.96.122:11037 --- INVITE sip:90018006273...@central.cipher.pe SIP/2.0 Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport Max-Forwards: 70 Contact: sip:488@181.64.96.122:11037 To: 90018006273999sip:90018006273...@central.cipher.pe From: 488sip:4...@central.cipher.pe;tag=93cce179 Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 56015 Content-Length: 235 v=0 o=- 8 2 IN IP4 192.168.7.33 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.7.33 t=0 0 m=audio 2424 RTP/AVP 0 8 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=alt:1 1 : hC2wRjti 7Lt7EhaI 192.168.7.33 2424 a=sendrecv - --- (12 headers 10 lines) --- Sending to 181.64.96.122:11037 (NAT) Using INVITE request as basis request - ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. Found peer '488' for '488' from 181.64.96.122:11037 --- Reliably Transmitting (no NAT) to 181.64.96.122:11037 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;received=181.64.96.122;rport=11037 From: 488sip:4...@central.cipher.pe;tag=93cce179 To: 90018006273999sip:90018006273...@central.cipher.pe;tag=as77d2f824 Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. CSeq: 1 INVITE Server: Asterisk PBX 1.8.11.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0a1fded4 Content-Length: 0 Scheduling destruction of SIP dialog 'ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.' in 11648 ms (Method: INVITE) --- SIP read from UDP:181.64.96.122:11037 --- ACK sip:90018006273...@central.cipher.pe SIP/2.0 Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport To: 90018006273999sip:90018006273...@central.cipher.pe;tag=as77d2f824 From: 488sip:4...@central.cipher.pe;tag=93cce179 Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. CSeq: 1 ACK Content-Length: 0 - --- (7 headers 0 lines) --- --- SIP read from UDP:181.64.96.122:11037 --- INVITE sip:90018006273...@central.cipher.pe SIP/2.0 Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport Max-Forwards: 70 Contact: sip:488@181.64.96.122:11037 To: 90018006273999sip:90018006273...@central.cipher.pe From: 488sip:4...@central.cipher.pe;tag=93cce179 Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls
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 problem lots of us here have it working that way (of course you have to take care of security fail2ban etc ) On the phone side you might have to use stun but it depends on the firewall also you should set the phone to send a nat keep alive each 30 seconds (asterisk also sends a options packet to keep the nat open but doesn't always work ok ) -Original Message- From: Danny Dias ing.diasda...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 27 Apr 2012 10:22:38 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls
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 List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk + Phones behind different Nat Firewalls -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints and server-side DND (do not disturb)
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: [asterisk-users] hints and server-side DND (do not disturb) Hi, Currently I'm using hints to determine SIP presence. As I understand it, a SIP extension can be labeled as busy, ringing, etc, based on a channel status. So a channel MUST be present. If it isn't then the extension is considered to be available. If my statement is correct then is there a way to set the extesnion as busy even if there's no channel associated with this extension? eg. when an extension sets server-side DND (Do Not Disturb), it actually sets a boolean value in astdb. Whenever asterisk tries to route a call to this extension, it first checks this value. Obviously, there's no way I can use hints in this scenario, or is there? Is it possible to somehow create a dummy channel whenever an extension sets server-side DND (custom context) and delete it whenever it unsets it? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Routing premature media to the calling channel
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 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Routing premature media to the calling channel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming musiconhold via mpg123
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Streaming musiconhold via mpg123 At my wits end with this, and can't proceed any further so I'm hoping someone has seen this and can assist. I can not get streaming musiconhold to work with Asterisk. My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is CentOS 5.7. When I call the musiconhold class (default for example) I get nothing but silence. I've exhausted my troubleshooting capabilities at this point, I've tried everything I can think of to include: - a newer version of mpg123, I went with the latest version - verified I could play an MP3 file by itself in Asterisk by using the MP3Player application What does not work, is if I use the mpg123 application for musiconhold to play a standalone file or a streaming source. I seem to be missing something and I just can't quite put a finger on it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming musiconhold via mpg123
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: Stephen Brown stephen.brow...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 21 Feb 2012 20:04:29 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Streaming musiconhold via mpg123 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reading second rdnis
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 in the sip header as second divert Is there anyway I could easily extract the second divert header Asterisk 1.8 Thanks, Israel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
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 List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: and...@telesip.net, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] conferenced transfers No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. -- Technical Support http://www.telesip.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive
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 -Original Message- From: Jonas Kellens jonas.kell...@telenet.be Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 06 Feb 2012 12:19:18 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read digits during recording / DTMF in conference?
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 - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] read digits during recording / DTMF in conference? Hi, I want to create a system for incoming calls where, under some circumstances, callers get routed straight to voicemail (or some other means of recording a message) but if they enter a valid extension number then the recorded message would be abandoned and they'd be diverted to the extension number they entered. I realise this can be done with the voicemail app with operator=yes but the problem with this is that the caller has to press 0 while the announcement is being played. If they're too slow and recording has started, they've missed the opportunity. So I played around with ConfBridge and a couple of call files, just to see if I could get it to work. It's a bit convoluted but the idea is that the caller gets silently put into a conference, then two call files make asterisk silently connect to other calls into the same conference, with one doing the recording and the other using Read() to collect digits. If I just had the caller and one of the other calls in the conference (the one doing Read()) then this worked - Read() managed to read the DTMF digits and assign them to a variable. However, when the 'recording' call is also in the conference, the 'read' call can no longer recognise the DTMF digits. To test, I made the 'read' call play a sound before calling Read() and I could hear this being played so the call was definitely there. However, regardless of the number of digits I pressed, Read() didn't notice any of them, even if I introduced a delay so that the other channels were quiet before the call to Read(). I realise this might seem a bit like a mad solution but can anyone else think of a way to get Asterisk to read (and react to) DTMF digits during a recording? This is with Asterisk 1.8.7. -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP hardphone with dual gigabit ethernet ports
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 Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SIP hardphone with dual gigabit ethernet ports Hi, I'm looking for a SIP hardphone with 2 network interfaces at 1 Gbps. All the ones I've seen only have dual 10/100Mbps ethernet ports (eg. Grandstream products). Any suggestions? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech recognition in asterisk using google voice API
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 Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Speech recognition in asterisk using google voice API -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to play wav files in 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 Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Not able to play wav files in asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit # of inbound calls on SIP trunk
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
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Limit # of inbound calls on SIP trunk
Un-top-posting...
On Mon, 19 Dec 2011, Douglas Mortensen wrote:
I have a system with FreePBX, and as far as I can tell it does not
provide a means to limit the number of simultaneous inbound calls on a
SIP trunk. Therefore I suspect that I’ll need to do some manual
dialplan manipulation.
On Mon, 19 Dec 2011, Steve Edwards wrote:
The GROUP() and GROUP_COUNT() functions and the GOTOIF() application
should do the trick.
On Tue, 20 Dec 2011, Douglas Mortensen wrote:
Excellent. Do you think these functions would enable me to create rules
based on both the concurrent # of inbound and/or outbound calls, or only
total # of concurrent calls (agnostic to call direction being inbound
vs. outbound)?
If you want a call to be a member of multiple groups, you have to play
with the category parameter.
exten = *,n,set(GROUP()=incoming)
exten = *,n,set(GROUP(incoming)=no)
exten = *,n,set(GROUP(incoming)=yes)
exten = *,n,set(GROUP()=outgoing)
exten = *,n,set(GROUP(outgoing)=no)
exten = *,n,set(GROUP(outgoing)=yes)
exten = *,n,verbose(incoming count = ${GROUP_COUNT(incoming)})
exten = *,n,verbose(outgoing count = ${GROUP_COUNT(outgoing)})
exten = *,n,verbose(incoming category count =
${GROUP_COUNT(yes@incoming)})
exten = *,n,verbose(outgoing category count =
${GROUP_COUNT(yes@outgoing)})
exten = *,n,verbose(group list is ${GROUP_LIST()})
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Forcing a CODEC
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 08:50:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Forcing a CODEC That's one of the uses of the SIP_CODEC dialplan variable. Just set it in the context or the sip.conf or users.conf. In your particular case, just set up a specific context for the IAX calls [iax-in] Exten = _X.,1,Set(SIP_CODEC=G722) Exten = _X.,n,answer() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius Sent: Tuesday, November 15, 2011 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Forcing a CODEC Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No call progress sounds
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 Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] No call progress sounds I recently switched to a PRI from analog lines. For reasons out of my control, my vendor had problems getting the PRI to interface so they set it to use T1-CAS instead. The lines are working just fine for inbound and outbound calls, except I get no call progress sounds. So no ring, busy, etc. When you place an outbound call, you just have dead air until the called party picks up. If it is a busy number, you have no way to know as it just sits with dead air until you give up and hang up. I have two lines for faxing stripped out of the T1 from the router and those have all proper audio, so this may very well be something misconfigured on my end, but I can't figure out what. I am using a Sangoma A101 card for the interface and running Asterisk 1.6.2.13. Anyone have any idea what I need to change to get the call progress audio? Thanks! -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
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 If you are going to use a T1 you should return the MP124 you have no need for that -Original Message- From: Nick Khamis sym...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 4 Nov 2011 09:07:11 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID from Direct from Telco I realized there was an error in my last post. I meant analog gateway plugged into and FXO port. DIDs must start somwhere. And I am under the impression that the telcos are the one that have control over that? Therefore, we would first need an analog gateway plugged into an FXO, before being able to go through the T1s and media servers? Your insight is greatly appreciated. Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
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 DID's on 1 T1 You need a digital gateway for connecting to a T1 Did you check if your provider will give you a T1 or maybe they could provide you a sip trunk which will save you on the hardware -Original Message- From: Nick Khamis sym...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 3 Nov 2011 22:10:31 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID from Direct from Telco Fair enough, In regards to the the diagram discussed earlier: Telco Lines - Gateway T1 - SIP Proxy - Media Servers - Customer I understand that a T1 Gateway that has 480 channels, can handle up to 240 calls. That is more than enough for the Gateway T1 - SIP Proxy part of the diagram. I just want to make terribly sure I understand the Telco Lines - Gateway T1. If the Gateway T1 plugs into only 1 FXS port, is that FXS port only capable of handling 2 channels, i.e., one call? Thanks in Advnace, Nick. On Thu, Nov 3, 2011 at 9:24 PM, James Sharp ja...@fivecats.org wrote: On 11/03/2011 09:16 PM, Nick Khamis wrote: Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Routing Florida numbers up to Canada would get you charged LD per minute fees. You can go with a provider like Level 3 or Global Crossing and they can hand you a T1 circuit that has DIDs from many different areas in the US. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
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 Discussionasterisk-users@lists.digium.com Reply-To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID from Direct from Telco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Executing outbound dial number twice
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 Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Executing outbound dial number twice Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604@sipphones:1] Set(SIP/4773-0003e920, CALLERID(num)=2066604) in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604@sipphones:2] Dial(SIP/4773-0003e920, SIP/att/xxx,80) in new stack Can you please help? Thanks, Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way voice with IVR
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 Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] one way voice with IVR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call pickup
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 Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call pickup On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some NOTIFY to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without monitoring the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bounty for ASTERISK-17474 streamingMusicOnHold bug
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 Hamburg l...@solvent-llc.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 26 Sep 2011 00:36:28 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] bounty for ASTERISK-17474 streaming MusicOnHold bug Danny Nicholas wrote: 2. Don't know if moving to 10.x would help you, but since that is still considered beta, that's probably not an option anyhow. Yup, not really an option for me. I actually use this system daily and don't want to muck around with 10.0 just yet. 3. My understanding is that bounties need to be posted on the asterisk-dev list. Fair enough, I couldn't find that info - can anyone else confirm this? I don't want to go barging into the dev list looking like a fool. 4. With those caveats, have you tried this: Copy the load_module and unload_module routines from res_timing_pthread.c to res_timing_dahdi.c (you'll probably need some includes [..] Hehe - no I definitely haven't tried that. That's a bit above my pay grade for now. I was hoping to find a more formal fix for this. Still clinging onto the idea that with a decent bounty put together, someone who knows the code well enough would be able to fix this. The fact that it WORKS GREAT until the first 'moh reload' suggests to me that it might be a relatively easy bug to squash. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms rebooting themselves
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: asterisk-users-boun...@lists.digium.com Date: Tue, 30 Aug 2011 15:26:59 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycoms rebooting themselves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering machine answers after pickup a phone.
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 Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Answering machine answers after pickup a phone. Hi! I'm sorry that I have misundertood your question, didn't read it carefully enough. So you have your asterisk and your phone conntected to the same incoming line. Maybe you can try with to detect an answered call with BackGroundDetect() exten = s,1,Answer() exten = s,n,BackGroundDetect(silence/10) exten = s,n,Voicemail(1234) exten = talk,1,HangUp() I can't try it for your setup with a POTS line, but I think this might work, especially when you tune the time values for BackGroundDetect(). Quote of the manual: --- SNIP --- -= Info about application 'BackgroundDetect' =- [Synopsis] Background a file with talk detect [Description] BackgroundDetect(filename[|sil[|min|[max]]]): Plays back a given filename, waiting for interruption from a given digit (the digit must start the beginning of a valid extension, or it will be ignored). During the playback of the file, audio is monitored in the receive direction, and if a period of non-silence which is greater than 'min' ms yet less than 'max' ms is followed by silence for at least 'sil' ms then the audio playback is aborted and processing jumps to the 'talk' extension if available. If unspecified, sil, min, and max default to 1000, 100, and infinity respectively. --- SNAP --- Hope this helps. regards, Ruben Am 05.08.2011 10:59, schrieb Jorge Barreiro: Hi again, thanks for your answer, but it didn't solve the problem. That Dial command returns inmediately, so I don't even have the delay. I'll try to explain myself better. The PBX has only one FXO card, connected to the PSTN. There is no other phones connected to the PBX nor SIP extensions. There are analog phones connected to the same PSTN. What I try to do is that, when there is an incoming call from the ouside, if someone answers on a phone, then the PBX won't answer. Thanks. O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu: Hi, your concept using Wait() won't work here. Try it like this: [incoming] exten = s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s exten = s,n,BackGround(wellcome-message) exten = s,n,Voicemail(1234) exten = #,1,Hangup() So, of you answer the call within 30s, you'll get the call on your phone. After 30s, the Voicemail will answer the phone. regards, Ruben Am 04.08.2011 21:39, schrieb Jorge Barreiro: Hello, I'm configuring an Asterisk PBX to use as an answering machine. I have a FXO card connected to the line, and other analog telephones connected to the same line. The PBX answers and redirects you to the voicemail after a delay. The problem is that even if I pickup any analog phone in the line before the PBX does, it answers after the delay anyway. And I couldn't find how to prevent this, or even if this is supposed to happen. My FXO card is a cheap X100P (source of problems, I know), and I'm using the Asterisk version included in Debian Squeeze (1.6.2.9). My dial plan looks like this: [incoming] exten = s,1,Wait(8) exten = s,2,Answer exten = s,3,BackGround(wellcome-message) exten = s,4,Voicemail(1234) exten = #,1,Hangup I don't know if this is related, but I'm in Spain and I had to add: hanguponpolarityswitch=yes to the chan_dahdi.conf file so that asterisk detects the remote hangup. I also added: answeronpolarityswitch=yes but this didn't help. It seems to be used just to detect the answer when you are calling, not when receiving a call. I'd appreciate any help you could provide. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- jumping frog - Webhosting Housing Ruben Rögels Moltkestraße 24 79098 Freiburg Tel.: 0761 / 384 78 85 Web: http://www.jumping-frog.org/ eMail:
Re: [asterisk-users] Increasing volume ?
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 14:52:52 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Increasing volume ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] use ILBC installed from asterisk yumrepositories
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use ILBC installed from asterisk yum repositories -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveFax to G.711
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 - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ReceiveFax to G.711 On 06/27/2011 08:06 AM, Michael wrote: Controlling it through the sip.conf peers is sufficient for us for this case (because this particular provider doesn't support T.38 at all), but I think it would be a good idea to add the option to enable/disable T.38 from the dialplan. If I recall correctly, that's how callweaver worked at the time. In Asterisk trunk (soon to become Asterisk 1.10), there is an 'F' option to ReceiveFAX and SendFAX that forces audio mode FAX even if the channel is T.38 capable. That would do what you want. I posted a patch some time ago for Asterisk 1.8 to add the same ability, but it probably doesn't apply any more... the it's only a few lines though, it should be fairly easy to replicate in the Asterisk 1.8 version of res_fax.c Also, we just checked it, and since for that provider, we have other codecs in higher priorities (like GSM, for example) than G.711, G.711 was not chosen as the only codec, so the fax transmission failed. We can not prioritize G.711 over the other codecs in the sip.conf, for the obvious reasons, so for this, we need to do it in the dialplan. How can we do it? How are you going to determine that you need to force G.711 to be used? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Missed calls and groups
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 want recorded Regarding queues if you call agents using the specific channel driver like sip/200 then it works but if using the local channel driver there was a bug reported (as far I remember) that it didn't work (it might have been fixed) -Original Message- From: russ...@lls.lls.com (Russell Brown) Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 17 Jun 2011 12:30:21 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Missed calls and groups Is there a SIP header I can set (for Snom and Yealink phones if that's relevant) or any other mechanism to tell a phone to ignore a particular call from it's missed call list? I have bits of the dialplan that ring groups of phones eg: exten = 200,1,Dial(Sip/112SIP/113SIP/114) and I don't want such calls being recorded by the phone as a missed call. Calls to the specific phone I do want displayed so just disabling the Missed Calls feature on the phone doesn't cut the mustard. Ideas? (I'd also want this to work with Queues but let's see about the basics first) -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callerid issue
-Original Message-
From: Steve Totaro stot...@asteriskhelpdesk.com
Sender: asterisk-users-boun...@lists.digium.com
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 - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Callerid issue
On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906 275
44578907 277
44578908 354
I need to setup the callerid with this extensions . for example whenever I
am dial from 354 extension callerID will show 44578908.
fro this scenarion I need logical dialplan because I have 100 extenstions ,
so 100 extentions should be have different extensions.
Below dialplan is for sequence callerid and extesions. like 101 to 199
should callerid is going 44578900 to 44578999 .
exten =_0X,1,NoOp(Int exten:${CALLERID(num)})
exten =_0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten =_0X,3,NoOp(Ext ident:${outgoing_ident})
exten =_0X,4,Set(CALLERID(name)=${outgoing_ident})
exten =_0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten =_0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten =_0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten =_0X,9,Hangup
--
Best Regards,
Mahesh Katta
BUZZWORKS Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
Are you using SIP phones? I assume you are but just checking.
Look into sip.conf
For each phone, add callerid=Joe Smith 1551212 no quotes in sip.conf
Be sure the telco allows it, some times they will only allow the BTN.
I have run into troubles with toll free callerid. Everything made
sense because the problem was calling other toll free numbers and who
pays. Good point. Especially because ANI can be manipulated in
Asterisk as well.
I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the common numbers and only sending four digits.
I guess if you get huge there could be duplicates of the last four.
Just set it in sip.conf
Thanks,
Steve T
Thanks
Steve Totaro
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Re: [asterisk-users] please help
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] please help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi command not available
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 asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi command not available -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so
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 -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 13 May 2011 15:17:24 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] With what options is asterisk compiled in rpm's
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 things installed from rpm and install whatever else I need from source (or roll my own rpm for those) Thanks, Israel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no ringback tone on outgoing call PRI line
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 - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] no ringback tone on outgoing call PRI line -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP Trigger on incoming call request
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 - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] TCP Trigger on incoming call request -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
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 Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 and new the command: exten = _X., 4, Wait, 2
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 asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.8 and new the command: exten = _X., 4, Wait, 2 OK Dears; Is the exten = _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent? I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me: Executing [9615806234@a2billing:1] Answer(SIP/gwsshihabuddinkw-0014, ) in new stack [Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application 'Wait,2' for extension (a2billing, 9615806234, 2) == Spawn extension (a2billing, 9615806234, 2) exited non-zero on 'SIP/gwsshihabuddinkw-0014' Now, my investigations: The extensions.conf: [a2billing] exten = _X.,1,Answer exten = _X.,2,Wait,2 exten = _X.,3,DeadAGI,a2billing.php exten = _X.,4,Wait,2 exten = _X.,5,Hangup ; From the other side: I did installations for Star2Billing version 1.9, I copied the a2billing.conf to the /etc/, also I enabled the manager.conf with port 5038. I copied a2billing.php and the lib to the agi-bin directory and I ran chmod +x for a2billing.php to make sure it is executable. And my php packages are: [root@Call-Bilal asterisk]# rpm -qa | grep php php-pgsql-5.2.9-2.fc10.i386 php-pear-1.7.2-2.fc10.noarch php-common-5.2.9-2.fc10.i386 php-pdo-5.2.9-2.fc10.i386 php-mbstring-5.2.9-2.fc10.i386 php-cli-5.2.9-2.fc10.i386 php-5.2.9-2.fc10.i386 php-mysql-5.2.9-2.fc10.i386 php-imap-5.2.9-2.fc10.i386 php-gd-5.2.9-2.fc10.i386 Note: I am able to place a normal call frome extension to extension but using a2billing, no success. Any help? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 and new the command: exten = _X., 4, Wait, 2
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 Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.8 and new the command: exten = _X., 4, Wait, 2 OK Dears; Is the exten = _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent? I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me: Executing [9615806234@a2billing:1] Answer(SIP/gwsshihabuddinkw-0014, ) in new stack [Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application 'Wait,2' for extension (a2billing, 9615806234, 2) == Spawn extension (a2billing, 9615806234, 2) exited non-zero on 'SIP/gwsshihabuddinkw-0014' Now, my investigations: The extensions.conf: [a2billing] exten = _X.,1,Answer exten = _X.,2,Wait,2 exten = _X.,3,DeadAGI,a2billing.php exten = _X.,4,Wait,2 exten = _X.,5,Hangup ; From the other side: I did installations for Star2Billing version 1.9, I copied the a2billing.conf to the /etc/, also I enabled the manager.conf with port 5038. I copied a2billing.php and the lib to the agi-bin directory and I ran chmod +x for a2billing.php to make sure it is executable. And my php packages are: [root@Call-Bilal asterisk]# rpm -qa | grep php php-pgsql-5.2.9-2.fc10.i386 php-pear-1.7.2-2.fc10.noarch php-common-5.2.9-2.fc10.i386 php-pdo-5.2.9-2.fc10.i386 php-mbstring-5.2.9-2.fc10.i386 php-cli-5.2.9-2.fc10.i386 php-5.2.9-2.fc10.i386 php-mysql-5.2.9-2.fc10.i386 php-imap-5.2.9-2.fc10.i386 php-gd-5.2.9-2.fc10.i386 Note: I am able to place a normal call frome extension to extension but using a2billing, no success. Any help? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spa8000 t38 faxing
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 they never end up speaking to each other These were fixed in the 2102 according to the release notes Well now I hope I could get someone at cisco to look at it because I have more than a dozen 8000's Thanks for your help -Original Message- From: Larry Moore lmo...@starwon.com.au Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 27 Mar 2011 11:26:35 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] spa8000 t38 faxing Perhaps this will help. I have a SPA8800 which has 4 x FXS 4 x FXO ports. It took me some time to produce a working configuration. In Asterisk I have the following where 904 is the extension of the fax-modem and itsp is you VoIP Service Provider. sip.conf [general] . . faxdetect=no t38pt_udptl=yes,redundancy,maxdatagram=400 . . [904] ; Cisco SPA8800 FXS Port 4 ; Analogue FAX Modem attached type=friend defaultuser=904 secret=secret call-limit=2 qualify=yes canreinvite=no directmedia=no directrtpsetup=no ignoresdpversion=yes transport=udp,tcp host=dynamic context=your_context faxdetect=no . . [itsp] . . faxdetect=yes ignoresdpversion=yes . . I am including information from my SPA8800 for one of the FXS ports I have a Fax Modem attached to, the key to getting it to work I believe is the FAX Tone Detect Mode. Audio Configuration Preferred Codec: G711a Second Preferred Codec: Unspecified Third Preferred Codec: UnspecifiedUse Pref Codec Only: no Silence Supp Enable: yes Silence Threshold: medium G729a Enable: no Echo Canc Enable: yes G723 Enable: no Echo Canc Adapt Enable: yes G726-16 Enable: no Echo Supp Enable: yes G726-24 Enable: no FAX CED Detect Enable: yes G726-32 Enable: no FAX CNG Detect Enable: yes G726-40 Enable: no FAX Passthru Codec: G711a DTMF Process INFO: yes FAX Codec Symmetric: yes DTMF Process AVT: yes FAX Passthru Method: ReINVITE DTMF Tx Method: AVT DTMF Tx Mode: Strict DTMF Tx Strict Hold Off Time: 40FAX Process NSE: no Hook Flash Tx Method: None FAX Disable ECAN: no Release Unused Codec: yes FAX Enable T38: yes FAX T38 Redundancy: 1 FAX Tone Detect Mode: callee only Symmetric RTP: yes Supplementary Service Settings CW Setting: noBlock CID Setting: no Block ANC Setting: noDND Setting: no CID Setting: yes CWCID Setting: yes Dist Ring Setting: yesSecure Call Setting: no Message Waiting: noAccept Media Loopback Request: automatic Media Loopback Mode: sourceMedia Loopback Type: media Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Filtering on from caller id
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: asterisk-users-boun...@lists.digium.com Date: Fri, 25 Mar 2011 09:14:45 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Filtering on from caller id -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call being rejected
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 Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call being rejected -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover Routing
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 hangup reason because I have some providers sending congestion even for unallocated numbers -Original Message- From: Rizwan Hisham rizwanhas...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 1 Mar 2011 17:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Failover Routing -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.2.4: SIP dialogs not killed?
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 -Original Message- From: Terry Wilson twil...@digium.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 28 Feb 2011 16:32:51 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8.2.4: SIP dialogs not killed? Hi, I'm wondering if this is normal asterisk behaviour: asterisk*CLI sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 10.12.0.2(None) 3c2f7ff2975e-wp 0x0 (nothing)No Rx: PUBLISHguest 10.12.0.2(None) 3c2f7f21b71b-9q 0x0 (nothing)No Rx: PUBLISHguest 10.12.0.2(None) 3c2f98afd6d8-6f 0x0 (nothing)No Rx: PUBLISHguest 10.12.0.2(None) 3c2e34be8647-jz 0x0 (nothing)No Rx: PUBLISHguest [...] 10.12.0.2(None) 3c298beb68fb-km 0x0 (nothing)No Rx: PUBLISHguest 10.12.0.2(None) 3c2e36b6bbfd-37 0x0 (nothing)No Rx: PUBLISHguest 10.12.0.2(None) 3c2e60ed3a98-4c 0x0 (nothing)No Rx: PUBLISHguest 10.12.0.2(None) 3c2f83a42bf2-2y 0x0 (nothing)No Rx: PUBLISHguest 10.12.0.2(None) 3c299ad4975e-fo 0x0 (nothing)No Rx: PUBLISHguest 173 active SIP dialogs asterisk*CLI asterisk*CLI core show uptime System uptime: 1 day, 19 hours, 59 minutes, 47 seconds Last reload: 1 day, 4 hours, 23 minutes, 23 seconds 21 sip peers [Monitored: 13 online, 6 offline Unmonitored: 2 online, 0 offline] Any idea what this might cause or how I could find out more about this calls? These aren't calls, they are SIP PUBLISH transactions (presence information, etc.). If you want to show active calls you would use core show channels not sip show channels. Terry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
