similar to: question about CDR

hello list, i have asterisk 1.4.43 installed and i want to configure the auto-answer exten => 506,1,SIPAddHeader("Call-Info:\; answer-after=0") exten => 506,n,Dial(SIP/105) when i call the 506 the SIP/105 still ringing, i have snom 320 and i have set the Auto Answer Indication: on i test with Dial and page() but the issue still the same any help please -------------- next

I appreciate the help I've been given so far. The issue I face is that the data I'm working with has 53000 rows, so in calculating distance, finding all recids that fall within 2km and summing the population, etc. - a) takes too long and b) have no sense of progress. Below is a loop that reads each recid one at a time, calculates the distance and identifies the recids that fall within 2

Hi all, I've been struggling learning R and need to turn to the list again. I've got a dataset (comma-delimited file) with the following fields: recid, latitude, longitude, population, dwelling and age. For each observation, I'd like to calculate the total number of people and dwellings and average age within 2 k.m. Distance could be Euclidean, however, a proper distance

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:

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:

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

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 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

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).

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

Greetings ... Sorry for cross posting, but I have seen this problem on both lists ... Okay, I am not sure if this is a problem only on RedHat using the rpm, because I tried only with rpm installations ... Install Samba3 from rom on RedHat 9. Configure and join domain, kewl. Test winbind with wbinfo -u and -g, also kewl, but no answer from getent passwd ... put in winbind in

Hello, i have asterisk 1.4 installed and i want to use click to call in order to do an outbound call if there is any php code in order to do this operation thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110519/417ac394/attachment.htm>

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 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

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