similar to: app_dial Flag

Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten =>

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.000000000 +0100 --- app_dial.c 2009-12-01 09:29:19.000000000 +0100

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch

Looks like half a patch has been applied to app_dial in cvs head could someone with commit rights fix it. Thanks Chris

Wondering if anybody else is experiencing this: Using June 21st CVS Call made internally from one Polycom IP600 to another. Core dump with the last message in log as: NOTICE[17426]: app_dial.c:681 dial_exec: Unable to create channel of type 'SIP' Happens a couple of times a day. No, I haven't done any backtracing, verbose logging, etc., (first thing in the morning, I promise) I

On Fri, 2004-05-07 at 16:30, anthm@lists.digium.com wrote: > Update of /usr/cvsroot/asterisk/apps > In directory mongoose.digium.com:/tmp/cvs-serv17955/apps > added D() parameter to app_dial to allow post connect dtmf stream to be sent using above call > +" 'D([digits])' -- Send DTMF digit string *after* called party has answered\n" > +"

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

Hi, I have configured tdm400p alone and it works fine for me. But now i tried to add 2 x100p in the same machine with the following configs, but asterisk refuses to load, can someone guide me what i have done wrong. Following r the configs and output. TIA, Azher >> ./asterisk -vvvvvc [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

Hi there, I have a TDM400P (1 x fxs, 1 x fxo) which I'm attempting to run on linux 2.6 (gentoo), without much success at the moment. I have previously had it working on a 2.4 installation, but when I moved to a new box and installed a 2.6-based system, it failed to work. In both cases I'm using whatever (libpri, zaptel, asterisk) is checked out by default (I assume that means HEAD)

This has got to be a stupid error I'm making... I have been experimenting with different hardware and software configurations before I decide what to use as a production platform. Up until just recently things were going well. But now it appears I'm unable to get access to my TDM400p from Asterisk. I know the TDM card works fine, used it in another machine where it performed flawlessly.

i pulgged the TDM400P to my computer and when turning on and dialing on zap the following message is shown : app_dial.c:960 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) == Everyone is busy/congested at this time (1:0/0/1) please help me thanks lot. __________________________________ Discover Yahoo! Have fun online with music videos, cool games, IM and more.

I've tried to ask this before but didn't want to waste a lot of bandwidth. However, I have to give the params in order to make the exact question understood. We have 2 ZAP FXO connected to two phone lines(2xX100P) 3 phones connected to a TDM400P with FXS modules Usually a caller will be on ZAP/2 and the first outbound line in the g1 group is on ZAP/1. There are two (main) call waiting

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME

If I have in my dialplan [AgentQ] exten => _XX.,1,Dial(Sip/{$exten},120,g) exten => _XX.,2,NoOP(here we are) where [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus: