similar to: auto-dail for ZAP channel, the application gets executed before the call attended

I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I have no ability to figure this out

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24

Hello. I'm trying to set TTS with Cepstral and Swift but can't get it to work. I get this error when testing it: -- <SIP/101-00000000> Playing 'welcome.gsm' (language 'es') -- Executing [702 at local-calls:3] Swift("SIP/101-00000000", "Hello this is ceptral") in new stack [Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to

Hi All, I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful. Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The call is initiated and that all works, my problem is that it does not wait for the line to be

Hello OK, in my case, there is only one samba server acting as PDC. On the PDC, it has a openldap server as backend. I have configured another server as the slave ldap server. slave ldap server will pool data by syncrepl. There are some spaces in samba/smbldap-tool that we can configure multiple ldap servers (or load balance by use of DNS) What happen if the PDC write data to the slave ldap

Hi, I'm sending the following message from port X to port 5060 of another box running Asterisk, and it is answering back to port X from port 5060. Shouldn't Asterisk use the Via header to find out where to answer, and in this case send its answer to port 4000? OPTIONS sip:192.168.0.103 SIP/2.0\r\n Via: SIP/2.0/UDP 192.168.1.130:4000;branch=0.0\r\n CSeq: 4711 OPTIONS\r\n\r\n Thanks,

Hello I Installed the Ceptral voices and Iam trying tu use the swift module with asterisk. But when I start it show: [app_swift.so]May 15 17:53:09 WARNING[18876]: loader.c:325 __load_resource: libswift.so.4: cannot open shared object file: No such file or directory May 15 17:53:09 WARNING[18876]: loader.c:554 load_modules: Loading module app_swift.so failed! Il looked for that library

dear all, can anybody tell me, how to backup my data on samba?do you have a script for backup incremental? i don't understand to user rsync thank you Cyd ____________________________________________________________________________________ Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/features_spam.html

Hi List; [incoming] include => parkedcalls exten=103,1,Dial(SIP/Bob,,tT) exten=104,1,Dial(SIP/Charlie,,tT) When we use tT and when we use t alone or T alone, I know this for call parking, but I do not know what the tT does? Regards Bilal ____________________________________________________________________________________ Sucker-punch spam with award-winning protection. Try the free

I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] <Dovid> u serious ? [02:15] *** mog

I have a home network with my dial up connection on the windows 98 machine. Since I have installed SAMBA and configured the computers to see each other, I can no longer pull up a website on the windows 98 machine. The problem is that I have assigned a permanent IP address for the Intranet. I have to use DHCP for the dail up connection. Therefore the IP address that I assign to Windows machine

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this..... - -- James Finstrom -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp fW2JPZdYl/uxG1ziUwYnHGo= =QPbv -----END PGP

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

Hello, I'm trying to get the following situation. Someone calls an application on one of our asterisk server. In this application the caller will call a SIP client. (with the command Dial) The Sip client is connected with another asterisk server. (see below) Caller --> asterisk01 (incoming server) --> asterisk00 (outbound server) --> SIP client (X-lite) Do anybody now how

Hi, I have a question about the dial command. Is the following scenario possible. 1) - Our asterisk server had a successful outbound call. - Our asterisk server has to call another caller and when answered asterisk has to connect this call to the another outbound call. My first question is , do I have to this with a DIAL command, of a MEETME command? (A) -

Hi i installed ceptral and i want to test it with asterisk can u plz tell me if i was wrong here>> ?? exten => 2,1,Answer exten => 2,2,system(/opt/swift/bin/swift "hello world") exten=> 2,3,Hangup() Mahmoud Badran ATSI Tel: +20 2 607 8917

Hi Thomas, Really Thanks for your advice. I have tried your suggestions but problems still remain. My findings is that some Windows, regardless of version, can map to Samba through dial-up networking but some just cannot. I try to find the difference between these Windows but failed. Regards, Neil Thomas Cameron wrote: > Neil - > > The problem you are having is possibly from one of

IPSwitchBoard Version 0.69 has just been released; it is available for FREE: <http://mambo.thorben.dk> Download here Release notes: * Record calls by right clicking any extension button, you can have several recordings at the same time. * Bug fixes The recordings will be placed as a single wav file on the Asterisk server in the folder: /var/spool/asterisk/monitor the name of the