similar to: FXS - TDM400 - No dial tone

I have been trying for some time now to make the hook flash work on the FXS port. I am using Asterisk 1.4.10.1 with zaptel 1.4.4. When I "manually" flash the hook I can manage to find the duration to put a call on hold. However when pushing the flash button it never works. The phone's flashtime seems to be too short. I tried to set a shorter flashtime in the zapata.conf file, but it

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

G'day all, A weird problem has just started happening with my * server, and I've googled myself stupid looking for help. My system is latest CVS, a TDM400P with two FXS modules, two X101Ps, and VoIP connections (SIP, SCCP & IAX2). My wife complained about her extension (analogue, on one of the TDM400 ports) being dead -- no dialtone. I had to shut down *, then unload and reload the

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

Hello, I seem to have the Freeswan IPSEC tunnel working between my two sites, but I am still having a problem that looks to be because of something I have configured wrong in my shorewall setup.. I have a LEAF Oxygen < 1.9 heavily modifed firewall setup.. Using FreeSwan 1.91, and Kernel 2.4.8. Modified to use IPTables and standard Debian network/interfaces. I am also using Shorewall

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -------------------------- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed Details:

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",

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

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Has anyone implemented "hot seating" in any neat way? This where people can log in to any phone in the company and have their calls/voicemail come to that particular handset.....

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack NOTICE[245776]: File app_dial.c, Line 698

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

Can someone refresh my memory how to backlist caller ID in asterisk 1.8? I had it working in ver. 1.4 but in 1.8 it changed. -- Joseph

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

Hi, I''ve started getting an error I can''t debug. The error doesn''t generate anything in my development log. A breakpoint set right at the beggining of the controller method does not get triggered. It happens on a form submission. I can make changes to the object model from with the console, but not on this form submission. I thought maybe it was routing, because I have

Hello Asterisk users, Can someone thwack me with a clue stick please? I am following the Asterisk TFOT book Dial() example trying to get the limit and announcements to work as per below. These settings seem to have no effect. There are no warning messages after 4 minutes or every 30 secs thereafter and the call lasts longer than 5 minutes. gunner*CLI> show dialplan [ Context