similar to: Is there a default dial plan that is not in extention.conf?

Displaying 20 results from an estimated 1000 matches similar to: "Is there a default dial plan that is not in extention.conf?"

2010 Jul 10
1
How can get user inputs from called party after dial?
Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks, Eyal -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Jul 20
0
No subject
one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in
2007 Sep 11
2
Asterisk 1.4.11, res_features.so, SegFault
Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps
2006 Nov 16
2
installing asterisk for Ubuntu Synaptic
I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>
2007 Mar 15
2
A200 card problem
Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or
2011 Feb 18
2
pbx_ael.so: undefined symbol: ast_compile_ael2
Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI> module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47]
2010 Aug 05
1
Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.
2003 Apr 03
4
Password Aging Policies - SAMBA
I am new to SAMBA. I am using it with LINUX and I love it! Nice change from you know who. I am sorry if this is a lame question, but I am setting up a Primary Domain Controller and a Backup Domain Controller. All the clients are Windows 2K or XP. I need the clients to reset their passwords every 30 days. I am having a hard time locating a procedure to set this feature. LINUX has 'chage'
2010 May 12
1
problems with unicall
Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type
2009 May 03
2
Asterisk not starting up due to database problems
When I try and start asterisk I get the following, however I have commented out the data the connections in res_mysql.conf and res_pgsql.conf. I am not sure therefore why I am getting these errors. Do I have to change something else to turn this off? Thanks Asterisk 1.4.21.2~dfsg-3, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk
2020 Jan 07
2
unexpected UPS status
This is part UPS and part nut question. I had a (intentional) power outage. The UPS/nut properly shut the system down when battery was down to 20%. It ran for only 18 minutes rather than the expected 40 (see later). After it charged to 37% I removed power from the UPS. After 4-5 minutes the system shut down. However, when I restored power to the UPS the machine powered up immediately (no delay).
2008 Feb 21
3
How to get a clean, basic configuration?
Hello I'm using a standard Asterisk install with default settings, and when I run "reload", I see that Asterisk fetches configuration information from a lot more sources than just my extensions.conf and sip.conf. For instance: -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to
2020 Aug 19
4
Eaton 5E 1100i USB UPS - failed to claim USB device error
Hi BACKGROUND INFO:I have a new Eaton 5E 1100i USB UPS which their website claims is supported on Linux. They don't provide software for it for Linux, only for Windows.Referring links: https://www.eaton.com/za/en-gb/catalog/backup-power-ups-surge-it-power-distribution/Eaton_5E_UPS.html https://www.eaton.com/za/en-gb/skuPage.5E1100IUSB.specifications.html It was manufactured in May 2020
2012 Jan 09
2
create table in mysql using asterisk
Hi, I try to create a new table using MYSQL command in asterisk. This is what i write: Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600" ("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40), "number_in_line" int(2), PRIMARY KEY("id")") and this is the warning that i get in the cli: app_addon_sql_mysql.c:383
2012 Jan 05
1
STOP loading extensions.ael
How do I stop loading extensions.ael dial plan? I'm only using extension.conf. -- Joseph
2019 Sep 06
5
"battery is low" logged
I got around to unplugging my UPS and I found it doesn't last nearly as long as estimated. Is the "upsmon[1446]: UPS desktop_ups at localhost battery is low" logged when "battery.runtime:" = "battery.runtime.low: 120"? Does any make better batteries that don't go down faster when they get old? battery.charge: 100 battery.charge.low: 10
2007 Feb 23
2
Any way to get rid of AEL created contexts?
"show dialplan" keeps showing contexts created by AEL. I tried deleting /etc/asterisk/extensions.ael but kept getting these messages in the Asterisk log: Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Feb 14 21:39:53 WARNING[6074] pbx.c: Requested contexts didn't get merged Is there any way to delete or
2019 Jul 19
1
USB UPS problem
Maybe a kernel USB issue, but people on this list may have encountered this issue. I am testing a new UPS (EATON 5E 1500iUSB) and am having a problem with the USB connection failing. My old UPS (EATON Ellipse MAX 1500) works just fine with the latest f30. Both devices identify as ID 0463:ffff MGE UPS Systems UPS To narrow the focus I booted recent live CDs on a PC that has no other USB devices
2011 Oct 27
5
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 at 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 at sipphones:2] Dial("SIP/4773-0003e920",
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang