search for: 6004

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=ye...

Hello, Suppose a dialplan such as: exten => 6004,1,Answer exten => 6004,n,Wait(1) exten => 6004,n,SayDigits(1) exten => 6004,n,Festival(This is a test of Festival) exten => 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004 at internal:1] Answer("SIP/505-00000004", &quot...

Hi Guys, Just new to Asterisk and am completely stumped. I have created two accounts as instructed. Please see below for the config of the user accounts. [Peter] type=friend host=IP address disallow=all allow=ulaw allow=alaw callerid=Peter <6004> secret=XXXXXXX context=default port=9060 nat=force_rport,comedia deny=0.0.0.0 permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0 When attempting to register there appears to be something not allowing the authentication of the client against Asterisk. I am getti...

Hi guys I have installed Xen 4 on Debian Lenny 5.0. I have a physical Windows 2003 Server and I wanna convert this server on a virtual server... I try to use XenConvert but without sucsess... Somebody can help??? Thanks Gilberto Nunes TI Selbetti Gestão de Documentos Telefone: +55 (47) 3441-6004 Celular: +55 (47) 8861-6672 _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

...am = "/home/xxx/scripts/mongrel_rails_start 6003" stop program = "/home/xxx/scripts/mongrel_rails_stop 6003" if totalmem > 50.0 MB for 5 cycles then restart if failed port 6003 protocol http with timeout 10 seconds then restart group mongrel #6004 check process mongrel-6004 with pidfile /home/xxx/sshare/app/log/mongrel.6004.pid start program = "/home/xxx/scripts/mongrel_rails_start 6004" stop program = "/home/xxx/scripts/mongrel_rails_stop 6004" if totalmem > 50.0 MB for 5 cycles then restart if failed...

...~~~~~~~~~~~~~~~~~~~~~~~~~~~~ == Using SIP RTP CoS mark 5 -- Called SIP/6003 -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 answered SIP/6004-00000000 -- Channel SIP/6004-00000000 joined 'simple_bridge' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07> -- Channel SIP/6003-00000001 joined 'simple_bridge' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07> > Bridge 2a01fb30-96e2-48b7-baaa-c2f...

...uration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=friend regexten=2101 username=2101 host=dynamic disallow=all ;allow=gsm allow=ulaw extensions.conf exerpt: exten => 6003,1,Dial(SIP/1003,15) exten => 6003,2,Voicemail(u1003...

...dateFormat = D.M.YA ; M-D-Y in any order (5 chars max) bindaddr = 192.168.1.37 ; asterisk box. port = 2000 ; listen on port 2000 (Skinny, default) debug = 10 [devices] type = ata description = Bedroom tzoffset = 0 autologin = 6004 device => ATA0016c7a0e0d4 type = ata description = Office tzoffset = 0 autologin = 6000 speeddial = 6001,6001,6001@internal speeddial = 6004,6004,6004@internal device => ATA16c7a0e0d401 type = ata description = LivingRoom tzoffset = 0 auto...

...; feature activation. Default is 500 [featuremap] blindxfer => # ; Blind transfer ;disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record (a.k.a. Touch Monitor) atxfer => * ; A users.conf: [6004] fullname = Analog User 4 secret = 6004 email = cid_number = zapchan = 4 context = numberplan-local hasvoicemail = yes hasdirectory = yes hassip = no hasiax = no hasmanager = no callwaiting = no threewaycalling = no mailbox = 6004 hasagent = no group = 2

hi, exten =>6000,1,dial(SIP/6000,15,tr) exten =>6002,1,dial(SIP/6002,15,tr) exten =>6004,1,dial(SIP/6004,15,tr) exten =>6006,1,dial(SIP/6006,15,tr) exten =>6008,1,chanspy(SIP/6006 | wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI> -...

Hi All, I am using "asterisk-11.12.0" version and I am trying to setup secure call (TLS + SRTP) between two extensions and while making a call, I got following error *CLI> == Using SIP RTP CoS mark 5 -- Executing [6004 at from-office:1] Dial("SIP/6003-00000000", "SIP/6004,20") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/6004 SSL certificate ok == Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure [Nov 2 21:20:05] WAR...

...l voicemail and listen to the message just fine which I guess rules out voicemailbox misconfiguration. The strange thing is that both extensions and mailboxes are configured exactly the same : in extensions.conf : exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 6004,1,Voicemail(u6004) in voicemail.conf : 1234 => 4242,Test mailbox,jim@jipo.com 6004 => 4242;Other test mailbox,jim@jipo.com I don't understand why these two seemingly identical configuration yield different results. I guess that I must have missed something that was included in the examp...

...s delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. 2nd question: using a grandstream phone & asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? 3rd question: can someone give me some "starter hints" to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving...

...work. However you can't use g1, you could use zap/*. > Is there a way to define a button for CAPI/[contr0] which lights up, no > matter what MSN is used? There was some discussion about CAPI, I never played around with CAPI before so I can't tell you. > When making a call from SIP/6004 to SIP/6005, the SIP/6004 button shows > ->6005 but the 6005 Button shows <- -. CallerID issue. Upgrade to latest CVS, or use the o option in dial. > Sometimes, when using a SIP button to originate, transfer or hangup a > call, the button disappears. When the mouse is moved over, the...

...6002,1,Dial(SIP/6002) ;exten => 6002,2,Hangup() exten => 6002,1,Set(_SIPSRTP_CRYPTO=enable) exten => 6002,2,Dial(SIP/${EXTEN}) ;exten => 6003,1,Dial(SIP/6003) ;exten => 6003,2,Hangup() exten => 6003,1,Set(_SIPSRTP_CRYPTO=enable) exten => 6003,2,Dial(SIP/${EXTEN}) ;exten => 6004,1,Dial(SIP/6004) ;exten => 6004,2,Hangup() exten => 6004,1,Set(_SIPSRTP_CRYPTO=enable) exten => 6004,2,Dial(SIP/${EXTEN}) exten => 6005,1,Dial(SIP/6005) exten => 6005,2,Hangup() ;exten => 6005,1,Set(_SIPSRTP_CRYPTO=enable) ;exten => 6005,2,Dial(SIP/${EXTEN}) exten => 6006,...

...Just new to Asterisk and am completely stumped. I have created two > accounts as instructed. Please see below for the config of the user > accounts. > > [Peter] > type=friend > host=IP address > disallow=all > allow=ulaw > allow=alaw > callerid=Peter <6004> > secret=XXXXXXX > context=default > port=9060 > nat=force_rport,comedia > deny=0.0.0.0 > permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0 > Your phone is registering with the name 6004 and not Peter. You either need to change [Peter] to...

...> private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten => s,1,Dial(Local/6001@internal&Local/6004@internal,15,rt) > exten => s,2,Wait(1) > exten => > s,3,Dial(Local/6001@internal&Local/6003@internal&Local/6004@internal,20,rt) > exten => s,4,Voicemail(u810920) > exten => s,5,Hangup > exten => s,104,Voicemail(b810920) > exten => s,105,Hangup > &...

...---> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in the above setup I'm able to call from Asterisk to my CME and vice-versa. here is my problem: when I call from 6004 to my cme extension 4615, on 4615 I've configured noans timeout to 15 and then it goes to my unity express (cue) for voicemail so when I call my cme extension it rings for few seconds and then on my asterisk cli I see "500 Internal Server Error" back from my CCM IP and getting standar...

...always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal : SCCP/6002 State:Idle Watchers 0 6001 at internal : SCCP/6001 State:Idle Watchers 0 6000 at internal : SCCP/6000 State:Idle Watchers 0 6004 at internal : SIP/sgofferj State:Idle Watchers 0 6200 at internal : SCCP/6200 State:Unavailable Watchers 0 3000 at internal : gtalk/gtalk_account/ State:Idle Watchers 1 Funnily, the gtalk hint is the only one with a watcher although all hints are hooked in vario...

...d"; auth-nxdomain no; # conform to RFC1035 listen-on-v6 { any; }; tkey-gssapi-credential "DNS/selb.local"; tkey-domain "SELB.LOCAL"; }; Thanks for any help Gilberto Nunes Ferreira TI Selbetti Gest?o de Documentos Telefone: +55 (47) 3441-6004 Celular: +55 (47) 8861-6672 "Bendita a na??o cujo Deus ? o SENHOR!" 99 <><