similar to: Outband call from php script

I'm using asterisk as a sip client with a sip proxy server... I've made the pertinent extensions and I've configured the sip.conf correctly or I think so.. I'm using x-lite as a client and when I ring to a public telephone through proxy, the arriving sound it's perfect but the sound I send is very bad, they hear me like a robot and distorted. Anyone know what's the

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which

I found this tracing a bug when experimenting with a new sort() function using match.arg(). It was triggered because mosaicplot.default(.) has an argument called `sort' and calls itself the sort() function in which I was using match.arg()... Here is (input for) a small clean example : #### bad match.arg() // sys.function() bug : #### MM, 22.3.2002 callme <- function(a = 1, mm =

Dear my friends... I am still new in CentOS. I use CentOS 5 for my new server with cpanel. I have installed the cpanel. The steps were: 1. cd /tmp 2. wget http://layer1.cpanel.net/latest 3. sh latest The installation show a message that the cpanel installation was completed. But I can not see the cpanel login form on my server, either: "http://WWW.MYWEBSERVER.COM:2082" or:

Hi, I use Centos 4.4, latest rpm for OWM. OWM runs fine but i want to enable speedy, so after I enable it I get the error: Not a perl script at /dev/fd/9 line 1., referer: http://mywebserver/ speedy_backend[3633]: perl_parse error, referer: http://mywebserver/ speedy[3631]: Cannot spawn backend process, referer: http://mywebserver/ my speedy is installed from RPM.

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Sorry for this long message. The two first questions asks for general coding standard when creating packages, the third one asks if the .R files are concatenated in lexical order or not, the fourth question is a "how-to" question. All questions are related. If there is a reference where I can read about this please tell me, because then I could repost a more restricted set of questions.

Hi, I worked with puppet (< 0.25) back in 2008/2009. We were able to deploy 200 servers from scratch and manage them. It worked fine. I''m now with a new customer and I''m pushing Puppet (and I''m also back to puppet on a side project). We''re considering Puppet 2.6 to manage RHEL/CentOS 5 or 6 hosts. I''m "upgrading myself" to Puppet

I downloaded R v2.1.1 earlier this morning to compile under Fedora Core 4. It compiled without incident, but 'make check' failed. Below is the relevant part of its report. Is this a known problem? I used a locally compiled version of GCC v4.0.0 that reports [kent at d89h102 R-2.1.1]$ gcc -v Using built-in specs. Target: i686-pc-linux-gnu Configured with: ../gcc-4.0.0/configure

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file

Hi all, A new version of DIAX (0.9.8a) is ready to be downloaded from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.8a: - unconditional autoanswer or based on CallerID (user configurable); - use any Ericsson/SonyEricsson GSM/PCS to control DIAX (feedback on the phone display) through Bluetooth (or serial cable). You do not even

Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax

Hi all, I have a small problem to execute Asterisk Commands in Asterisk Manager using PHP. I am able to run all Asterisk Manager command but the problem is comming with asterisk command. here is the code i am trying to run. <?php $socket = fsockopen("localhost","5038", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket,

Hello Using pxelinux for so long time i don't want to use pxegrub to PXE boot solaris 10 on x86 . So i would like to use pxelinux to do this jobs . I use ipappend and mboot.c32 however i don't success to have the kernel argument appended with BOOTIF : here is what i use : LABEL solaris MENU DEFAULT MENU LABEL Solaris Jumpstart KERNEL mboot.c32 APPEND multiboot kernel/unix

A while back, I asked about using Asterisk in a medical environment where the task is to write a program that connects to a phone and sends a message like: Hello Mrs. Jones. How are you doing today? Press 1 if you're OK. Press 2 if you need help. Or start talking, and your message will be passed to a person. After connecting and sending the sound file, the program would

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

Guys, is anybody using PHP sockets to connect to the Manager and send command like "show voicemail users" for example or any other? My question is, how to parse the return info in a way that can be shown back to the user via web (discard all the manager responses not needed)? Thx in advance for any help you can provide.

I recently began using the curl cmd to do an external callerid lookup on my own customer database. I've noticed certain lookups will cause a crash and not show anything in the messages file or the console. The curl command is connecting to an external webserver which has a oracle db connection. The file its hitting is PHP and does a very simply lookup showing the text like "C1234 Bobs

Hello, My Cisco phone has software: Boot Load: PC030300 Ver: 3.2(7.0) And I want to upgrade it to SIP 6.0 Is it possible or I have to upgrade to ealier then 6.0 and then to 6.0 ? bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031121/7331dff5/attachment.htm