similar to: Marco and Realtime Extension Problem

Dear All, Sorry to be posting again. I have solved my problem. The problem is that when exiting from the macro, the priority number is still in effect. For example, priority 1 is at the start before entering macro after the macro the priorty will be 2. Since there isn't any other dialplan command, the switch statement would be search for a priority 2 in the Realtime extensions table. One

Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my Asterisk 1.2.1 box. Also when compiling Asterisk 1.2.5 and tried to run it with the Asterisk-addon,

Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406

Dear All, It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a

Suppose I have one agent login into two different queue and there are calls waiting in both queues. If the calls in one queue has higher call prority (set QUEUE_PRO to higher value) than the calls in other queue, will the agent get the higher prority call first or the QUEUE_PRO has no effect? We have an Asterisk server( 1.2.17 with CentOS) running as ACD. We are having problem using weight option

FYI, just pushed: >From 322ff984a39d53422b772bfeb8f69e7c648da8c0 Mon Sep 17 00:00:00 2001 From: Jim Meyering <meyering at redhat.com> Date: Fri, 14 Aug 2009 21:01:48 +0200 Subject: [PATCH libguestfs] build: avoid "make sytnax-check" failure * daemon/configure.ac: Change a leading TAB to 8 spaces. --- daemon/configure.ac | 2 +- 1 files changed, 1 insertions(+), 1

I apologise for writing again about the problem with using stepAIC + multinom, but I think the reason why I had it in the first place is perhaps there may be a bug in either stepAIC or multinom. Just to repeat the problem, I have 126 variables and 99 cases. I don't know if the large number of variables could be the problem. Of couse the reason for doing a stepwise method is to reduce this

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

Hi, I''m trying to figure out how to use Linux QoS. Default setting has three queues (bands) and should prioretise outgoing tarffic based on TOS field. I try to test that by flooding Ethernet interface by netperf or iptraf and running ping -f with -Q and without Q. -Q doesn''t affect ping results, it suffers anyway. It seems that I don''t understand something. I verified

Hello, I created this setup, DSL------LINUX ROUTER-------ASTERISK Linux acts as router and forwards packets only 512M and AMD 1599.987 MHz Asterisk 512M AMD 2000 MHz When I ssh to linux router during the call and execute any command that requires cpu , then sound gets choppy. Simple test would be establish a call and start "du /" on the router. The same applies to asterisk box.

Hi All, I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL:

I''d like to use :through to create a web of associations like: class Thing < ActiveRecord::Base has_many :child_things, :through => :thing_thing has_many :parent_things, :through => :thing_thing, :some_other_option? end class ThingThing < ActiveRecord::Base belongs_to :thing belongs_to :child_thing, :class_name => ''Thing'', :foreign_key =>

Hi, I want to prevent Asterisk from sending the h extension across to the SIP provider or to prevent it from hitting the script at all. The SIP Provider does not know what to do with the h extensions once it receives it. My SIP Provider takes all digits and forwards them off to a softswitch for processing. Everytime a call hangs up, it complains about running AGI scripts on hungup

I have wondershaper to limit my upload at 400kilobits (my line is 600kbps). I do a lot of torrent seeding and I dont want my pings killed when I''m uploading so I set low prority source ports as follows (by the way, I have bittornet to only use ports 6881-6910): NOPRIOPORTSRC="6881 6882 6883 6884 6885 6886 6887 6888 6889 6890 6891 6892 6893 6894 6895 6896 6897 6898 6899 6900 6901

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line connected. I am new in Linux and Asterisk, my steps are theese: 1. Install CentOS 4.4 (basic instalation). 2. Command line: yum -y update yum install gcc kernel-devel bison openssl-devel yum install openssl-devel 3. Download the source: wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz

Hi, sorry to bother you guys with a simple sytnax question; i have a loop of objects taking place (ie, for page in @pages....xxxxxxx....end) and a link associated to each pages so that in the end it looks like this: page1 (link) page2 (link) page3 (link) . . . page n (link) (all of this done by putting a simple ''link to'' in the for loop.) now i need to seperate two pages

Hello Using Asterisk 1.4.20 and Zaptel 1.4.3-1, I notice that Asterisk sends FXO calls to a context named "numberplan-local" that is not mentionned in my configuration file, which prevents incoming calls to be successfull: ======= /etc/asterisk/zapata.conf ================== [trunkgroups] [channels] ;Send incoming calls to this context in extensions.conf context=from_fxo