similar to: Asterisk repository: asterisk14-addons-mysql

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

please can someone send to me his files like zaptel & zapta if he si using TE110P thank you

Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan ######### [root at localhost ~]# yum update asterisk18* -x asterisknow-version Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * base:

It seems that asterisk-addons and one or more of Digium's licensed modules such as res_fax_digium have a conflict that doesn't seem to be documented anywhere I can find. In a nutshell, asterisk14-addons-core has a fake provide for asterisk-gplonly : # # core subpackage # %package core Summary: Asterisk-addons core package. Group: Utilities/System Provides: asterisk-gplonly Provides:

Hello. I have a TE212 configured in E1 mode. This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured): cat /proc/zaptel/2 Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RED NOTOPEN 25 TE2/0/1/1 Clear 26 TE2/0/1/2 Clear 27 TE2/0/1/3 Clear 28 TE2/0/1/4 Clear 29 TE2/0/1/5 Clear 30 TE2/0/1/6 Clear

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

Hello all, I would like to share with you an article [1] we have issued last week (sorry, currently only in Romanian language - we plan to provide an English version soon). This article is describing a method to be used for obtaining the maximum number of SIP simultaneous calls an Asterisk server could process safely (meaning no errors/maintain control of the machine and without RTP frame drops)

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in "maxretry" in jail.conf For example, I get an email saying: "The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK." when "maxretry = 5" in jail.conf Perhaps someone else is experiencing this or has resolved it,

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

I'm build with the usage of the tcltk/tcltk2 package a gui which is started with rscript.exe. At this gui the user is able to open and close plot windows. Now I'm looking for a way that a r function is called when a grDevices::windows window is closed. Is there a way to attach a callback function or so, which would be triggert if the user click at close of the grDevices::window? My

I'm build with the usage of the tcltk/tcltk2 package a gui which is started with rscript.exe. At this gui the user is able to open and close plot windows. Now I'm looking for a way that a r function is called when a grDevices::windows window is closed. Is there a way to attach a callback function or so, which would be triggert if the user click at close of the grDevices::window? My

In my series of articles about Asterisk 1.4, I've now arrived to the new jitter buffer that enhances voice quality for those of you using Asterisk as a PSTN gateway. Please read http://www.voip-forum.com/category/asterisk/asterisk14/ /O

I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but

hi: my company has two dns servers, but samba4 internal dns forward can only forward to 1 server. to workarround it i run dnsmasq as dns cache at 127.0.0.1 and let samba4 dns forward to 127.0.0.1. my smb.conf looks like: dns forwarder = 127.0.0.1 interfaces = 10.1.1.1 bind interfaces only = Yes with the setting samba won't bind to 127.0.0.1, so dnsmasq can use 127.0.0.1 to do the work.

Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

Hi I am trying to use usrmgr.exe in order to manage users and groups on my samba server PDC (passdb backend = tdbsam ) . I have the following strange behavior of usrmgr.exe : * when I launch usrmgr.exe from a user account that is part of the Domain Admins group or that explicitly has the SeAddUsersPrivilege privilege, I can see the list of users and groups , and create a new user, BUT when I

This link (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. Does anyone know where to find that upgrade package? If it doesn't yet exist, what is the process for upgrading?

Hello, I'm writing here hoping to have a hint from Asterisk/Digium packager maintainer, Jason Parker (of course that any other's opinion is welcomed). We have installed an asterisk machine using Asterisk and Digium repos. Unfortunately we have found that an Astribank could not be connected due to the lack of the firmware files May 18 14:19:22 tenora1 kernel: usb 1-6: new high speed USB

Hi, I'm hunting a bug in 3.0.x (tested on 3.0.10 and 3.0.20). Users can create and modfy files, but cannot delete them. The logs show [2005/09/21 20:48:14, 0, pid=18388, effective(4019, 412), real(4019, 0)] tdb/tdbutil.c:tdb_log(767) tdb(/srv/physik.fu-berlin.de/data/.samba/cluster1-test/cache/locking.tdb): expand_file ftruncate to 8192 failed (Permission denied) [2005/09/21 20:48:15, 0,