similar to: Carrying context from one server to another?

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

Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: root at Sangoma-Testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? Thanks! -------------- next

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

Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by

I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8

Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with "Asterisk <Unknown>" caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server

Hello, is there anyone who can point me to correct information ? Following http://pbxinaflash.com/forum/showthread.php?t=9042 and http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR does not result in a working environment for me. Any feedback appreciated. Kind regards, Jonas. -------- Original Message -------- Subject: [asterisk-users] Asterisk 1.6.2.10 & CDR

Hi, I am doing failover routing based on 2 dial commands. First route sends back 4xx response and I don't want it to try 2nd route when it is 4xx response. Can we do failover routing based on SIP 5xx response only ? Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110309/fe9d7bc7/attachment.htm>

Hello everybody! First my setup: I connect from Debian Lenny to Ubuntu Karmic with a command like: user1 at localserver:$ rsync -rtcve ssh user1 at remoteserver:/.../ /local/.../ (using default versions of ssh and rsync in the vendor repos, ssh with password authentication) As far as I understand if localserver got compromised an attacker could read the password and then get full access to

We have a multitenant Asterisk 1.4 installation for multiple small business, and we need to report how many calls a single business has active at one time. Is there a way to VIEW how many calls are up in a single context? (Or some other way to accomplish the same)? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

2016-06-01 19:36 GMT-03:00 <cpolish at surewest.net>: > On 2016-06-01 09:53, Sergio Belkin wrote: > > Hi folks, > > > > I have a problem, shame on me I feel as newbie, I cannot open Firefox > > though ssh. > > > > ssh server is 7.2 and ssh client (running Xorg) is Fedora 23. Firefox > does > > not open. I've tried a lot of methods, even

The desired result is that user A's phone rings; when he picks it up, user B is dialled, and user A's channel is connected to that. (This is to be a back-end for a web-based address book.)

Hi! I am using mainly Samba 3.5 on CentOS, and I was very pleased with idmap_rid backend for SID-to-RID mappings. But on Solaris 10, I can only use 3.6 because OpenCSW ships only 3.6. Problem is, things are changed and are not working as expected... Here is my config on RHEL Samba 3.5: [global] workgroup = WINDOMAIN realm = WINDOMAIN.LOCAL server string = localserver

AXvoice server hacked. Here are few working accounts USE XLITE to make calls.... Registrar/Proxy magnum.axvoice.com:9060 Free Sample account.... username=xMaxwellSmartx secret=thanksapache username=woodsy type=friend secret=haramikuttasala username=wumingzi type=friend secret=kickyourass Enjoy! B.R BaBa Jigger -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Hi ! I have a python script that create and move .call files to /var/spool/asterisk/outgoing Sometimes...(in this case after 500 successfull calls) Asterisk don?t make the calls and the .call files are in the "outgoing" forever... Any Ideas? I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior) In my python script I move .call files using ... import shutil

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards