similar to: [PATCH] allow announcements in app_dial

Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

I have a user, running CVS a/o 11/23/03, who has complained about "phantom" messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/asterisk/voicemail/default/XXXX/Old directory, and to my surprise the messages are indeed

Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31

Hello, Thanks to members of the list things changed on my installation (asterisk + digium E100P with of course an E1 line). Here is my zaptel.conf : - span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr - Output of zttool: - Alarms Span YEL/RED Digium Wildcard E100P E1/PRA Card 0 - [root@asterisk root]# head /proc/zaptel/1 Span 1: WCT1/0 "Digium

<P>hi Everyone,</P> <P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P> <P>*CLI> <BR>&nbsp; == D-Channel on span 1 up<BR>&nbsp;&nbsp;&nbsp; -- B-channel 1 successfully restarted on span 1<BR>&nbsp;&nbsp;&nbsp; --

Hi, I have tried using a raw print queue with no drivers and printed from the windows end. Still Word or other app being printed from dies and no print request appears in the servers logs. Any more ideas? Cheers, Mark. -----Original Message----- From: brucehohl@access-4-free.com [mailto:brucehohl@access-4-free.com] Sent: 21 May 2004 10:27 To: samba@lists.samba.org Cc:

Hello, I am trying to use a HFC-PCI (CCD/Billion/Asuscom 2BD0) card in TE mode to dial-in and out with ISDN. The problem is I can not get the card to dial out with a Zap channel. Dial-in is working. I am using bri-stuff 0.1.0-RC4 (but tried also RC3 and RC2k). I tried all combination of "immediate", "overlapdial", "pridialplan". I earlier also managed to dial out

Hi People, I have just built a Samba 3.0.2 server using Suse Standard Server 8 and Cups for printing. Setting up printers and printing works fine for all of the more modern printers, but older HP Laserjet 5N / 4 printers do not. When trying to print to them from Win 2k/XP the windows app that we're printing from just dies, or gives a "do you want to send error report to

On the /proc/zaptel it was lost interrupts, you haven't got any so that's good! On opencall.org there's a known issues link and that mentions some fax machines that have issues, might be worth a quick check there. I can receive from our fax machine this end, if you'd like I'll send you a test fax if you send me your fax number, if you can receive from us and tpc then it's

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Sorry to bore you more with the clock issue, but have you check /proc/zaptel/<span> to make sure it's not missing interrupts? There's also an option to record the audio for the fax, you could listen to that vs a recorded file that will receive correctly on a fax machine and see whether there is an obvious difference? (Good luck, that'll be really scraping the barrel!!) Does it

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

A nice flag in app_dial ? would be f and F to indicate weather to send flash to the fxo or fxs device in a bridged call. so if you have a pots line on an x100p bridged to a tdm400p and the pots line has call waiting you hear the call waiting signal 'f' would behave as normal and flash on the tdm400p and 'F' would send the flash on the x100p The flashing the tdm400p

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" > +"

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

I have back ported the 'r' feature for app Dial from 1.8.0 to 1.6.2.X. The link to the diff is below. http://files.bluecrow.net/asterisk/backports/1.6.2/asterisk-1.6.2.4-app_ dial-play-indications.diff I made the diff against 1.6.2.4 and later patched a 1.6.2.13 system. All hunks passed. I have this running on two production 1.6.2.X systems with no problems yet. The

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