similar to: Difference between Transfer and Dial applications

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was

Hello, I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to providers Cisco 2800 with VWIC-1MFT-E1 card. the same card runs fine with another E1 provider. TE420 led's lite green. Message type: RELEASE COMPLETE (90) < [08 02 80 ac] < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) < Ext: 1 Cause:

hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone number for few seconds, checks if it "hears" the ringing and hangs up the call. ** I use

Hi, I have IVR and when I press 1, asterisk calls my mobile phone. If my mobile phone is offline, asterisk transfers to asterisk voicemail. I'd like asterisk detects my mobile voicemail and if my mobile voicemail answers, asterisk transfers to asterisk voicemail. For that, I used AMD. So I have problems ! Asterisk detects answering machine everytime! How do I do please ? extensions.conf

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Hello All, I do have asterisk installed for a call centre with aheeva application and i would like to know how to configure the sound for the inbound calls and if there is any possibility for agent to receive a file with the phone number and name of clients: For your information there is no problem related to the outbound call An help would be appreciated Kind Regards Salah. --------------

Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, ------------------------------------------------------- Dario Quiroz (71) 9275-9080 gtalk: darioquiroz at gmail.com ------------------------------------------------------- -------------- next part -------------- An HTML attachment was

The vast majority of what I've done with Asterisk has been with the Grandstream GXP-2000's. These phones work great for us for everything *except* speaker quality is quite poor and appears to be half-duplex. So now that we've bought and are using 40 GXP-2000's we're doing some testing on other phones. I've bought a Polycom 301 & 501 as well as a Linksys SPA942. While

When inserting Ringing() before MeetMe()-conference picked up the call, everything works like a charm. I guess the PRI needed to see the ringing status before the call was answered. This is however never needed when dialing a SIP-extension or similar. I have also an update considering bad PRI b-channel numbering. It seems that only my first 15 channels actually work. Then our PBX tells Asterisk

Dear list, Is there any functionality in R that would allow me to parse config files? I have trie ??config and apropos('config') without succes, and also search the R package site. Mvh. Marie [[alternative HTML version deleted]]

Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from <IP-ADRRESS> When I google'ed this problem I can see other users also found this error (bug ?) But no-one

Dear list, I have a vector of elements which I want to combined each with each, but none with itself. For example, > v <- c("a", "b", "c") and I need a function 'combine' such that > combine(v) [[1]] [1] "a" "b" [[2]] [1] "a" "b" [[3]] [1] "b" "c" I am not very interested in the orders

OK! I'm sorry about this. As I wrote earlier I'm totally lost... but I will try to explain the problem in steps bellow, ok. 1. I installed the rpm's for samba, Version 2.0.2a-ssl I think this is the version distributed with redhat linux 7.0 2. Then I changed the parameters in the /etc/samba/smb.conf file, and in this file I added the folowing parameters. [global] netbios name

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

CentOS Errata and Security Advisory CESA-2009:0476 pango security update for CentOS 3 x86_64: https://rhn.redhat.com/errata/RHSA-2009-0476.html The following updated file has been uploaded and is currently syncing to the mirrors: x86_64: updates/x86_64/RPMS/pango-1.2.5-8.i386.rpm updates/x86_64/RPMS/pango-1.2.5-8.x86_64.rpm updates/x86_64/RPMS/pango-devel-1.2.5-8.x86_64.rpm source:

Hello, I'm not understanding how to use GOSUB_RESULT in U() option from Dial app (I'm using 1.6.1) My extensions.ael is : context mylocal { 2 => { Dial(SIP/7530,,U(mynotify)); HangUp(); }; 3 => { Dial(SIP/7531); HangUp(); }; }; macro mynotify () { GOSUB_RESULT=ABORT; }; I

Hi I have a problem using the package cluster on my binary data. I want to try mona at first. But i get the an error. hc<-read.table("all.txt", header=TRUE, sep="\t", row.names=1) srt(hc) `data.frame': 51 obs. of 59 variables: $ G1p : int 2 1 1 1 1 1 1 1 1 1 ... $ G1q : int 1 1 1 1 1 1 1 1 1 1 ... $ G2p : int 1 1 1 1 1 1 1 1 1 1 ... $ G2q : int 1 1 1 1 1 1