similar to: I can't figure out how to redirect a call to a trunk.

Displaying 20 results from an estimated 3000 matches similar to: "I can't figure out how to redirect a call to a trunk."

2008 May 13
1
getting multiple network interfaces to work in 3.2
I have a couple of Xen 3.1 servers running on Ubuntu 7.10 AMD64 Server machines. I decided to to upgrade these machines up to the latest release of Ubuntu (8.04 LTS) which upgrades Xen to version 3.2. I upgraded the first machine, installed the xen component. these machines all have two network interfaces one connected to the WAN and the other to the local LAN network. In a tutorial i read a
2004 Dec 05
1
Hardware PSTN Gateways?
I am thinking about setting up an asterisk PBX system for my company. But since I can't be at all the locations all the time I am setting up an automatic backup system where if the backup detects that the primay is down it takes over the IP so calls can be made once more. For this reason I want to setup a seperate HARDWARE PSTN Gateway. Are there any equiptment that can be plugged into
2010 Jul 12
1
My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?
Hi Everyone, I have done some php coding to come up with my own FollowME module for FreePBX. The need for this has some security considerations behind it. This is what my code does at core: $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing, pre_ring) VALUES
2005 Aug 10
0
tdm400p / outbound zap prob
I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack
2007 Nov 13
2
lvm over nbd?
I have a system with a large LVM VG partition. I was wondering if there is a way i could share the partition using nbd and have the nbd-client have access the LVM as if it was local. SYSTEM A: /dev/sda3 is a LVM partition and is assigned to VG volgroup1. I want to share /dev/sda3 via nbd-server SYSTEM B: receives A''s /dev/sda3 as /dev/nbd0. I want to access it as VG volgroup1. I am
2011 Sep 05
0
Followme generate ringing instead of MOH
Afternoon All, Is anyone aware of a way to generate ringing as opposed to starting music on hold for the party originating a call with followme? I'm assuming its doable as it looks like FreePBX users get the option (Not to say that FreePBX haven't got their own followme implementation though). Cheers Nick.
2007 Aug 17
1
swap partition and live migration.
when performing a live migration i know the contents of the ram is copied over two the new xen server, but how about the contents of the swap partition. currently I have the swap partition in a seperate loopback file from the root partition. if i want to do a live migration, do i have to give the new server access to the swap partition file along with the root partition file? of can i just
2009 Sep 07
0
Freepbx database followme disable/enable value
Hello, I am writing an AGI script to achieve the following - Users can Disable/Enable followme from their extension. They can also change the followme details from their extensions. I have looked at the follow me table for freepbx. I can't see the field for the values enabling/disable followme. Is this value stored in the database? -- Best Regards, James Mutuku Ndeti Agile Systems
2006 Nov 07
0
Follow Me problems
From: "Time Bandit" <timebandit001@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date: Tue, 7 Nov 2006 08:53:51 -0500 Subject: Re: [asterisk-users] Follow Me problems > Today we appear to have discovered our first bug. We have an extension > setup to "followme" by ringing that extension
2010 Jun 16
0
H323 Trunk Problem calling from Asterisk to Avaya PBX
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote: > Hi! > I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide > a conference bridge for an existing Avaya PBX. I have no control over the > Avaya system, but I am able to speak with the admin in charge when I need > stuff done. I am running all this in a VirtualBox
2006 Feb 09
0
re: voipjet -- Workaround if needed
Same thing here. I had this problem awhile ago and made this workaround. Going to another trunk does not work because they are answering and not sending a error code. If you are using AAH code then this waits 10 seconds on your Voip then times out and goes to PSTN. You can modify for your needs The pertinent line is 14 in macro-dialout-trunk I am going to clean it up and repost under my
2006 Feb 10
0
Half Solved - Fail over to Pri on VoIP connection failure
I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes
2011 Sep 02
0
No subject
crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.
2009 Aug 20
0
asterisk followme feature code
Hellos, I have using asterisk 1.2 and freepbx 2.3. I need users to disable and enable followme from there phones. I can't see any support for it. Is this possible/available.? I have googled and I can't get information on it -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer
2010 May 21
0
FollowMe dials numbers but can't confirm the call or hear anything
Trying to do a FollowMe test. When the extension is dialed, it dials my cellphone and my cell phone rings. But when I answer my cell phone it's just silence. When I press '1' on my cell phone, nothing happens. extensions.conf: exten => 140,1,FollowMe(mleonetti) followme.conf [general] featuredigittimeout=>5000 takecall=>1 declinecall=>2
2008 Jan 25
1
Problem with FollowMe
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as
2004 Jun 01
1
Feedback needed! FindMe/FollowMe FeatureSpec.
Hi Adam, I appreciate your feedback, and understand totally where you're coming from as far as the database perspective goes. For the first "draft" of the app, I think I'm going to let it default to using a conf file for two reasons. First, my setup currently does not utilize a database. I would like to move to that type of a setup in the future however. Secondly, seeing as
2008 Jan 22
1
Followme
I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which
2015 Mar 12
1
Realtime followme and channel variables
Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar field. I just need to pass a variable from the channel placing the call to the followme to the channel
2005 Sep 04
0
FW: OH323 with Asterisk@home - seems incomplete
Thank you (for spamming) - it was the clue I needed to push this through. Sorry it took me a while (and a google :-) ) to realize you'd addressed my initial query - basically, my loss. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Becker Sent: Wednesday, August 24, 2005 00:36 To: Asterisk Users