similar to: Dial plan questions

Is it possible to give a caller three goes at an extension number then hangup? exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Ringing(1) exten => s,4,NoOp(${CALLERID}) exten => s,5,SetMusicOnHold(random) exten => s,6,Background(silence/1) exten => s,7,Background(thank-you-for-calling) exten => s,8,Background(silence/1) exten =>

There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get that error: *CLI> load cdr_odbc.so Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Unable to load module cdr_odbc.so But the file is there... # ls -lag /usr/local/lib/libiodbc.so* lrwxrwxrwx 1 root

OK, this may seem to be an obvious question but where do I find the reference docs? I'm getting this error message: Timeout, but no rule 't' in context 'home' about this line: exten => 2201,1,Dial(${PHONES1},20,Ttm) I know the problem is with the 't' but I don't know what the parameters mean. I looking for a man page basically. -- Linux Home Automation

Hi List Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly phonegc:/home/samantha# asterisk -r Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Connected to Asterisk CVS-05 currently running on

Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] => (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql:

This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom root@localhost.localdomain Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: root@localhost.localdomain (Cron Daemon)

The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call

Hi, I've a Fritz card which was working fine, recently I changed hardware and my nightmare started. Now when I call someone through the chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I always get hungup. Can someone please give some help? Here are the logs: *CLI> -- CONNECT_IND ID=001 #0x0000 LEN=0049 Controller/PLCI/NCCI = 0x101 CIPValue

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

If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Hi, I have an F3000 phone that I am trying to register to asterisk. As far as I can tell I have everything in correct. Are there any little quirks I need to worry about? The phone has internet access, set it's time.. I can access the web config, but it just won't register with asterisk. I don't see anything meaningful in the full log.

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

(Apologies to those Toronto Asterisk Users' Group folks who have seen this message... I figured I'd have more success with a wider audience) I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on the wire. I wonder if anyone has done this before and therefore can validate whether or not the traffic I am

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm

I have IAX2 (FWD) partially working. I can place calls from my Asterisk box but I cam unable to receive them (comes back as busy). I have my firewall forwarding the udp ports 5060, 4569, 5036 and 10000 thru 20000 to my asterisk server. I think I have the firewall correctly setup as I can forward other services to their appropriate servers. I have no mail box on the one account (the one I'm

I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -