similar to: SV: Little Php question

> -----Original Message----- > From: Ronald [mailto:asterisk107@gmail.com] > Sent: 26 May 2005 10:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Little Php question > > > Hi > I'm trying to make a call from a local webpagee through my > xlite softphone > (xlite1) > BTW when I'm trying to do it through

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 again, Well - I didn't see beta8a-2.3.3 in custom dir. Will try. Also I tried to contact Sangoma - they are very fast to answer but main problem is time difference - it's 6 hours between Canada and Europe. Br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax:???? +45 70 25 73 74 Web: www.comx.dk

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,

Ok, I have tested with almost all versions both in 2.3.2--*-stable and 2.3.3-*-beta - I am getting same messages: May 9 10:55:26 WARNING[3961]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! and same Down state pb01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7

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:

Hello, Have you tried the wanpipe-beta8c-2.3.3.tgz release in the custom/2.3.3 dir on their FTP site? Also, have you contacted Sangoma for support? They are very responsive. I am using wanpipe-beta8a-2.3.3.tgz and it's been working great on my A104 for a week now. MATT--- -----Original Message----- From: Dmitry Zhukovski [mailto:DZH@comx.dk] Sent: Monday, May 09, 2005 5:20 AM To: Asterisk

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

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 ;

Hi! My mail server works in tandem: Dovecot (IMAP) + Postfix. How can I configure in Gmail: Outgoing email copied to the Sent folder using postfix without mail client? This question is for the mailing list for Postfix? Mail client itself copies outgoing messages in the folder Sent, but the messages are large, and the server is far away. And to transmit email twice for sending and copying is