similar to: Outbound Dial via SIP

Hi Everyone, I am building a script for a client and my imagecreatefromjpeg() is throwing a fatal exception, and dying. It seems as my server with CentOS 4, PHP 4.3.9 doesn't have >= GD-1.8 support or better. I did a phpinfo() and I can not find any reference to GD..............Any help on this would be well appricated -- Best Regards, Jon McCauley

Hi everyone I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments. Cheers Matt

Dear all, When I print out princomp's loading outputs, there is alwasy a section for "SS loading", "Proportional Var" and "Cumulative Var". Anybody can tell what they are for? Or anyone can direct me to some reference to read about? Any help will be highly appricated. Hongsheng [[alternative HTML version deleted]]

Hi everyone Can anyone help? x = c("Treatment", "Planned", "Actual", "Demographics", "Outcomes") for (i in x) { df <- paste(i)[, c("Reference_ID", "Arm_ID")] unique(df) -> uniq as.data.frame(table(uniq$Reference_ID)) -> y i.Arms<- paste(y$Var1, y$Freq, sep="_") Print_Message("Check for the Total No of

Hello, I recently installed the ruby on rails framework from this tutorial: http://www.hivelogic.com/narrative/articles/ruby_rails_lighttpd_mysql_tiger So, I installed: Ruby on Rails 1.1 Ruby 1.8.4 LightTPD 1.4.13 FastCGI 2.4.0 RubyGems 0.9.0 Readline 5.1 PCRE 6.6 FastCGI and MySQL bindings I figured out how to keep my ruby on rails up to date by using the command "gem update rails

Has anyone had any luck getting a ReplayTV DVR box to connect through an Asterisk box? Mine seems to dial just fine, but can't negotiate a connection. I am using: exten => _95380024,1,Dial(Zap/1/${EXTEN:1},120,d) exten => _95380024,2,Congestion I don't have any problems doing a fax though my system. For this setup, I am running a simple Digium developer's kit on a

hi, I have strange problems with HTB and several hundred classes flat structure i.e. root |--50kbps |--30kbps |--50kbps |--80kbps |--100kbps .... several hundred classes like this Ceil is the same as rate. The machine get no more than 2-3% average cpu(2.4Ghz pentium). What happens is that from time to time the traffic got "stalled". I tried numerous things to solve

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes

Dear All, I have two data like this : $cat main.csv name,id,memory,storage mohan,1,100.20,1.10 ram,1,200,100 kumar,1,400,50 xxx,1,100,40 aaa,1,800,45 mount,1,200,80 > main <- read.csv(file='main.csv',sep=',' , header=TRUE) > main name id memory storage 1 mohan 1 100.2 10 2 ram 1 200.0 100 3 kumar 1 400.0 50 4 xxx 1

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",

Hey, I have rather new installation of xen, dom0 is running ntpd and is perfectly in time. All domU''s are ~23s in future. Running ntpd on the domU''s seem to have no effect, /proc/sys/xen/independent_wallclock appears to have no effect. I''m running xen-unstable and 2.6.16-rc4 in AMD64 box. Any tip would be appricated. Thanks, -- ++ytti

I am running Samba 3.0.7 on Gentoo Linux. The server is configured for domain logins and roaming profiles, and there are about 40 Windows XP Pro SP2 clients that login to the domain. This has been setup and working perfectly since October, until this morning. Now, anytime a client tries to login, they get this error: "windows cannot connect to the domain, either because the domain

I'm following the instruction from Jo?o Amaro from the page http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html Everything went fine until I run the 'make' command under asterisk-oh323-0.6.5. I got the error message chan_oh323.c:5220: too many arguments to function `ast_channel_register' I have attached the error message. I'm running asterisk CVS

I'm trying to produce some enhancements to one of the applications, and am trying to use ast_request("zap", format, "pseudo") to create a new channel on /dev/zap/pseudo, which I can then bind to a zaptel conference and play a stream to it. I've been using as inspiration the Radio Repeater app, app_rpt.c, which uses this technique to play idents and announcements.

Hi List, Puppet Version: v2.7.13 Question - If I correctly setup multiple manifestdir''s in the puppet.conf file how can I import a node pp file located in a different directory tree to site.pp? My configuration information is below. In my puppet.conf file I have setup an environment called "stable", within this environment I have setup the manifestdir, modulepath and

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: