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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

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 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 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 I have two Asterisk Server: The first server "A", all phone are connected The Second server "B" only route call to a lot of SIP supplier the server A sent: ; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR exten => _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW) exten => _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt) exten =>

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info.

Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364xxxx (official number) 081169xxxx (Nddi Number) When i receive a call on the 081169xxxx, he don't use the extension. He use the 003318364xxxx extension. SIP Debug: <--- SIP read from

Hi We want optimize my extensions file conf on asterisk 11.4.0 : We have a big quantity of extensions, all are same "design": ; Destination: Gambia Type: Fixe exten => _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) exten => _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) exten => _00220X.,3,Set(CALLERID(all)=${NUMID}) exten =>

Hi i use this into my extension : exten => _00339xxxxxxxx,1,Set(foo=${SIP_HEADER(To)}) exten => _00339xxxxxxxx,2,Set(cut1=${CUT(foo,:,2)}) exten => _00339xxxxxxxx,3,Set(CLI=${CUT(cut1,>,1)}) exten => _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)}) exten => _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten =>

Hi A small question on Asterisk Manager. I use Perl Script for start a call: my $response = $astman->sendcommand( Action => 'Originate', Channel => 'SIP/ASTERISK/$Extension', Exten => '200', Context => 'MyContext',