similar to: click to call with php

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Hi, Le 12/03/2015 17:28, Salaheddine Elharit a ?crit : > hello list, > > i use the code below > > [macro-chanspy] > exten => s,1,Authenticate(${ARG1}) > exten => s,n,ChanSpy(SIP/${EXTEN:3},__dqs) Here you have a problem: ${EXTEN} value is s [...] Daniel

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

<<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>> Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf

Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

thank you so much it work you must add 1 like below [app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) best regards. 2015-03-11 19:48 GMT+00:00 Carlos Chavez <cursor at telecomabmex.com>: > On 3/11/15 12:48 PM, Salaheddine Elharit wrote: > >> hello list, >> >> i use

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ====ok inbound and outbound the calls between x-lite and snom300====> ok inbound and

Hello all I have a small issue with some mobiles numbers when I call these numbers using asterisk I have all the time answer machine. But when I call these numbers using my mobile or another phone there is no problem. Any help will be appreciated -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list i have configured astersik 1.4 with sip i have a question when i put in dial plan.conf exten => _0678922645.,1,Set(CALLERID(number)=520460587) exten => _0678922645 .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => _0678922645 .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)) exten => _067892264*5*,2,Hangup() i can not call my

Hello List, i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip extension is

hello list i have an IVR menu in asterisk 1.4 like below exten => 600,1,Ringing() exten => 600,n,Wait(2) exten => 600,n,Goto(home,s,1) [home] exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/) exten => s,n,Background(${sounds_path}music1) exten => s,n,Background(${sounds_path}music2) exten => s,n,Background(${sounds_path}music3) exten =>

Hello All, I have asterisk installed in our call center and i want to know how to do in order to save all the calls (inbound and outbound) if there is any tool Thanks in advance Kind Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110131/224a8492/attachment.htm>

Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! -------------- next part -------------- An

hello list, i use chanspy with the code below [app-chanspy] exten => _007.,1,Macro(user-callerid,) exten => _007.,n,Answer exten => _007.,n,Authenticate(1111) exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => _007.,n,Hangup i have a question related to chanspy i have created extension from 100 to 300 and i will give the permission with group of extension i want to use

Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 17-31 context = default group = 63 but when i add in channel 1-15 like: channel => 1-15,17-31 i receive all

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten => 506,1,Dial(SIP/223, 10) exten => 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

Hi, I'm trying to write a simple perl script that will run the following: Action: Originate Channel: local/xxx@callback/r/n Exten: 1234 Context: callback Priority: 1 Extensions.conf exten => 500,1,agi,callback.pl callback perl script: use Net::Telnet (); $mgrUSERNAME='fred'; $mgrSECRET='bloggs'; $server_ip='127.0.0.1'; $tn->print("Action: