similar to: Asterisk as an outbound call machine?

All, I am currently working with the AGI interface using PHP, I have it working to execute commands that do not require prompts to be played, but when I execute an AGI command to play a prompt or stream a prompt all i get is silence (although I can see in the log where it says the correct name for the prompt to be played). I have looked through the wiki and googled extensively, so if something

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Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

I am curious partly because it has occurred randomly in my asterisk system. How does one go about creating a .call file for placing a call between two extensions/phones? I know this has been mentioned and is probably in one of the wikis somewhere, but I am unsure exactally how to go about doing it. Can anyone point me in the right direction. Dan

Hey all, There's a website I help run that uses the Cassandra DB as its database. I notice that if I run the web server in SELinux permissive mode, the site works fine. But if I put it into enforcing mode, the site goes down with this error: Warning: require_once(/McFrazier/PhpBinaryCql/CqlClient.php): failed to open stream: Permission denied in

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

I''m working on porting a dynamic website done in PHP over to mod_ruby. I have everything set up and working nicely (I can make a .rhtml page, put ruby code between <% %> and it works etc), but when I try to use "Eruby.import" as a sort of analog to PHP''s "require_once()", I run into trouble. 1) eruby is in my cgi-bin 2) here''s a sample of

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537) I would like to do this, but I am having trouble figuring out how. I have

Hi Jeremy, An easy way to start troubleshooting these is to look at the audit logs and > see what SELInux is blocking. You have /McFrazier in the email.. if that's > off the root tree than unless you've set permissions to allow httpd to look > at tat folder, I bet that's one problem. > if you run ls -Z you can see the labels that are present on those folders, > that

Hello I am anirudh, I need help to write script to fetch detail like ips, speedlimit from mysql and add htb rules at the starting of the server. it is for a small isp i am working for. can any one help me out!, i had tried but fail to limit speed -- Anirudh Chowdary _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl

No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have apache2-mod_php installed? Rgds, Julius. > DO you have apache2-mod_php installed ? > > Which distro are you using ? > >> -----Original Message----- >> From: asterisk-users-bounces@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >> Julius Kidubuka >>

Does anyone have a Perl or PHP Icecast2 Status Page Parser? Thanks. -greg. __________________________________________________________________________ Greg J. Ogonowski VP Product Development ORBAN / CRL, Inc. 1525 Alvarado St. San Leandro, CA 94577 USA TEL +1 510 351-3500 FAX +1 510 351-0500 greg@orban.com http://www.orban.com

Hello again, /thanks to Peter for previous response. I've been digging around trying to find sample usage of XapianTermIterator/XapianPositionIterator in PHP. The idea is to code up a test case in PHP to perform snippet extraction (with a possible view to coding a pecl extension in C). I found a C++ sample, but that wasn't much help. I must be dense this morning though, since I

On Fri, Dec 05, 2014 at 10:16:10AM +0100, Thorsten Glaser wrote: > On Fri, 5 Dec 2014, Gene Cumm wrote: > > > If so: http://www.syslinux.org/wiki/index.php/Mboot.c32 > > I???m not permitted to edit either that page or its talk page. > Do I submit the content addition here, then? Euh, yes, that is an option. Hereby I volunteer for updating the wiki with updates that get to

Hi, I would like to know how to see which channels are used in my PRI E1 link from Asterisk to another locally-connected commercial PBX. If I run "dahdi show channels", I can see the used channels in the second column "extension" but only if it's an "incoming" call (ie. legacy PBX to Asterisk). If I dial from an Asterisk extension to an extension in the other

CentOS team... as is already bug reported and marked solved... as we await the upstream repair for this. It was reported that this was happening on CentOS 5. You likely already know, but it also happens on CentOS 4. For those unaware. It seems that SquirrelMail has an issue which allows mail to be sent out from one user on the system and it uses the from address of another user on the

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

So apparently 1.8.4 is soon forthcoming. We need testing against it. Could someone help out with that? I believe Ara already checked into some of the issues, do you know if those are resolved, Ara? -- David Heinemeier Hansson http://www.loudthinking.com -- Broadcasting Brain http://www.basecamphq.com -- Online project management http://www.backpackit.com -- Personal information manager

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.