similar to: Asterisk 1.6 and Username in Dial

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 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 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 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 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 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 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 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 all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw

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 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 =>

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

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 =>

Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers => mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI> realtime mysql status

Anybody have any experience with setting up dhcpd in failover mode between two servers? I set this up on a couple of servers, and it seems to be working, but I don't think it is working "right". It appears both servers are replying to all requests (which for renewals works okay because they both give the same address, but new requests get two different responses). I thought that

Hi Guys, In stream_decoder.c when assigning lpc restore function, only IA32 processor benefits from SS2 and SSE4.1 optimization. Shouldn't it be the case for x86_64 processor as well ? Thanks, -- Olivier TRISTAN uvi.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac-dev/attachments/20140311/1d49b5c2/attachment.htm

Hi folks, I have successfuly migrated a WinNT 4.0 Domain to a Debian server with Samba3+Ldap following the Samba-3 by Example guide from John H. Terpstra (an impressive good guide) and The Linux Samba-OpenLADP Howto from Jerome Tournier & Olivier Lemaire. The domain holds about 800 accounts. There are WinNT servers, WinXP and Win95 clients belonging to it. WinNT servers and WinXP clients

On Tue, Feb 24, 2015 at 11:56:50PM +0100, Olivier Mauras wrote: > > > On Tue, 2015-02-24 at 16:05 -0500, Laine Stump wrote: > > On 02/24/2015 03:37 PM, Olivier Mauras wrote: > > >> Hello, > > >> > > >> I've been trying to boot a VM with the rootfs being a 9P share from > > >> the host. The VM OS is centos 7. > > >>

Hi actually, i test a new Asterisk Server and i want add Mysql Realtime SIP. I read on the wiki: =================================================== Database Config put the following in res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = myuser dbpass = mypass dbport = 3306 Values in sip.conf or iax.conf like in older versions of * are no longer used. Database Table Lets