similar to: Little Php question

Hi I think you should have a look at the end of line - you are missing " :-) Br, dmirty -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Ronald Sendt: 26 May 2005 11:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Little Php question Hi I'm trying to make

Hi All, Been having problem using the AMI, i've got this PHP script: $socket = fsockopen("1.2.3.4","5038", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket, "UserName: amiadmin\r\n"); fputs($socket, "Secret: amiadminpassword\r\n\r\n"); fputs($socket, "Action: Command\r\n"); fputs($socket,

Hello, I want to recieve the output from astmanproxy in a php script. Is that possible ? I made a simple php script: <PRE> <?php $socket = fsockopen("127.0.0.1","1234", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket, "UserName: xxx\r\n"); fputs($socket, "Secret: xxx\r\n\r\n"); fputs($socket, "Action:

Hi,I met the same problem as this mail, http://www.mail-archive.com/asterisk-users@lists.digium.com/msg101451.html ******************************************************* Hello,I want to recieve the output from astmanproxy in a php script.Is that possible ?I made a simple php script:<PRE><?php$socket = fsockopen("127.0.0.1","1234", $errno, $errstr,

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, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call:

Hi all, One of the areas I am trying to research before I can confidently start deploying Asterisk is "Credit Card Machines". (PDQ / Streamline machines / any similar) I'm talking about the credit card swipe boxes at point of sale desks. I believe they dial out to the specific bank provider everytime a card is swiped. My question is: - Does anyone have any experience using

Forwarded from the SWIG mailing list, so we have a copy in our archives. Kevin -------- Original Message -------- Subject: Re: [Swig] Re: Object return problem Date: Sun, 09 Oct 2005 18:31:40 -0400 From: Kevin Smith <wxruby@qualitycode.com> To: Charlie Savage <cfis@interserv.com> CC: Swig@cs.uchicago.edu References: <4347277E.1030700@mindspring.com>

Ok so I've just built and installed a CVS (HEAD) version of asterisk on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples via "make samples". Everything seems to work except one thing. I'm trying to do the connect to the Digium IAX demo server portion of the demo (dial 500) and I just get the following messages. I am behind a NAT server and did NOT change

Hello, These are error messages I get when I try to call a number over CAPI channel. -- Executing SetCallerID("SIP/xlite1-3b80", "0") in new stack -- Executing Dial("SIP/xlite1-3b80", "CAPI/hfcpci/b17") in new stack > data = hfcpci/b17 > capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00,

Hello, I have the following setup: (*)<--->IP<--->Micronet 5012 H.323 box <---> POTS <---> PBX (Alcatel OmniPCX) Grand idea is to use the micronet's POTS interfaces to connect SIP phones to the PBX and to the PSTN. I think i even managed my way in the arcane and cryptic management interface of that appliance, but I am stuck against theese messages: -- Executing

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hello list, Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Thanks. VoipCrazy

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

The configuration for kphone in sip.conf: [177204] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid="Jane Smith" <5678> host=dynamic ;nat=yes ;

Chris helpfully reviewed the libguestfs ruby bindings. His findings are below. Rich. Date: Wed, 5 Jan 2011 16:36:42 -0500 From: Chris Lalancette Subject: Review of libguestfs ruby bindings Hey Rich, What follows is a quick review of the libguestfs ruby bindings. I hope you find it helpful. Overall directory structure - looks reasonable enough. One thing that you *could* do is remove

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been