similar to: Asterisk, Sent accountcode between 2 asterisk

Hi I have a problems with a new Asterisk Server, when i want call, i have: [Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160 handle_request_invite: Call from 'PHISIP000001' to extension '00420225352184' rejected because extension not found. but into my extensions.conf: exten => _00420X.,1,Set(CDR(CodeTier)=CZE) exten =>

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

Compiled the 3.0.0 version of the source code , then tried lli --use-mcjit irfile.txt On both windows and linux, I got: LLVM ERROR: Unknow object format. If I omit the -use-mcjit option, the command works well. It seems to me that something about MCJIT is broken in the 3.0.0 version. Also tried to initialize an ExecutionEngine from code, got errors like "Target does not support MC

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. I

Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use "internal E1" card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier

Dear all I have problem with calculate probability, I have data x1,...,x10, I want to calculate probability x11 given x1,...,x10 with two conditions. 1. x is normal 2. unknow distribution How I can do this. Many Thanks. Jumlong -- Jumlong Vongprasert Assist, Prof. Institute of Research and Development Ubon Ratchathani Rajabhat University Ubon Ratchathani

I am thinking of buying a Portec MV370 (single channel VoIP/GSM gateway) I am after feedback reports both good and otherwise please. Thanks, Michael

I have recently posted a message regarding hung SIP channels when using the Monitor() command. Well, I was wrong.   The channel hanging wasn't caused by the Monitor command. They also hang when they aren't monitored. The cause seems to be our PSTN gateway provider. When for some reason their PSTN gateway crashes or reboots (OK, this should happen, but anyway...) and RTP/SIP data stops

Hello, I've tried to integrate samba 3.0.1 and LDAP 2.1.23 using the guide provided from http://www.hilinski.net/samba/. While the ldap+samba user authentication seems to work fine, I can't join the Domain from a Windows 2000 Client. The Domain is found and Name/Password Credentials are asked. I enter root and password and I get an error: Login Failure: Unknow username or bad

Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that "soft hangup" should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command "iax2 show channels") IP-AM-PBX*CLI> iax2 show channels Channel

Hi On my FC3 box with asterisk 1.0.9....MusicOnHold is not working. It starts and stops immediately... An unknow option mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any

Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx at Accueil_HNO:2] BackGround("SIP/SIP000005-0000000c", "Fermeture") in new stack [Jun 16 07:35:06] WARNING[7634][C-00000006]: file.c:701 ast_openstream_full: File Fermeture does

Hi i want use the API on my asterisk 1.6, but i have a small problems : In extension, i start it : exten => _X.,3,AGI(My-Script.agi) The perl agi file are started without problems but i want get into this script a lot of variable: Type (SIP or IAX) src (from cdr) but that's don't work: use Asterisk::AGI; use lib "/var/lib/asterisk/agi-bin"; $AGI = new

Hi In Asterisk 1.6/realtime Mysql, we can't put a username/password in a Dial Command ?: 'Dial', 'SIP/Username:Password at MYPEER/${EXTEN},180,r' Thanks Olivier

Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten => 0532xx,1,Answer exten => 0532xx,2,MusicOnHold(Sound_1) exten => 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten => 0532xx,4,Hangup When i call to the number, i have the Music "Sound_1" but the SIP

Hi, I have a problem with Samba. I use Samba 4.1.17 (because my company's security policy deny Samba upgrade) on Debian. I created a Samba Active Directory Domain Cotroller with linux.local and a Windows Active Directory Domain Controller with windows.local domain (I used Windows Server 2012 R2). I am need to create a smb share in Linux and autheticate users from windows.local domain and need