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I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

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

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 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 -

Just a note to those of us with the SPA-3000 that have had a hard time with the 'volume' of calls from the PSTN (we're on the line they're on the PSTN). Anyway I change the parameter: PSTN To SPA Gain to 6 and the volume is fine (it's under the PSTN Line tab). I don't remember seeing this on the list so I thought I'd add it here. Guess I should add it to the wiki (a

I recently lost my setup (bad drive) and I'm now trying to get my setup back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an SPA-3000. I can call the phone extension, I can call from the phone on the SPA to other extensions and I can call out to the PSTN. What I can't do is to call from the PSTN to through the SPA to Asterisk. It rings twice then I get fast busy. I

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.

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

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 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

(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

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

I had my setup working properly under 1.2 and after a disk crash I decided that I wanted to try Asterisk 1.4. So far I can transfer between phones and I can dial out. What I can't get working is to get an SPA-3102 to 'send the calls' to Asterisk. I have the device added to the sip.conf file and it shows up in users and peers (but not in registry). Where do I start with debugging? --

Does the ${BLACKLIST()} function allow for values other than 1 to be returned and if so how can I use that is the AEL? Can I use the function in a switch statement? -- Linux Home Automation Neil Cherry ncherry@linuxha.com http://www.linuxha.com/ Main site http://linuxha.blogspot.com/ My HA Blog Author of: Linux Smart Homes For Dummies

I use Google Apps for my domain, hosted on a CentOS server. How could I setup my sendmail.mc to route Sendmail mails through the Gmail SMTP? Any help would be greatly appreciated. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20080628/ca90aa1f/attachment-0001.html>

Hi, This is a bit of-topic, but since it has to do with /tmp permissions here it goes. Anyway, I'm sorry in advance for posting this. I have this PHP script (simplificated here), called delete_tmp.php that basically calls external commands: <?php $session_file = '/tmp/sess_89765' system(''rm -f' . ' ' . $session_file); ?> delete_tmp.php file is owned by