search for: extenson

...160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register => 8001187e0:password at sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten => 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick

...e the idea of having each line button on the phone as a separate sip device. If I understand it right, each phone could have three extensions (one for each line.) This would be great since I could then use the dialplan to forward calls to the desired extension. I envision something like this: Extenson 101 - Company-A Extension 102 - Company B I've adjusted my dialplan to prepend a "hint" to the callerid so that I know which menu choice the caller picked: exten => 1,2,SetCIDName(A-Sales: ${CALLERIDNAME}) exten => 1,3,Macro(stdexten,101,SIP/ext101) exten => 2,2,SetCIDNam...

Hi, What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/a...

....0.5 on Gentoo (from portage) and when I try to reload my settings I get: == Unregistered 'mysql' CDR backend == Parsing '/etc/asterisk/cdr_mysql.conf': Not found (No such file or directory) My phone or SPA-3000 will not register with asterisk either. I've copied the Sip.conf; extenson.conf and iax.conf from old version to 1.0.5 -- #Joseph

Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, "Juki" wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to }

...xtensions.conf [outbound] exten=>5141234567,1,Dial(Zap/1/${EXTEN}) exten=>5141234567,2,Playback(vm-goodbye) exten=>t,1,Hangup =-=-= In /var/spool/asterisk/outgoing/1.call , the content of the file is: Channel: Zap/1/5141234567 MaxRetries: 3 RetryTime:300 WaitTme: 20 Context: outbound Extenson: 5141234567 Priority: 1 -=-=-= And the zap configurations in zapata.conf are: signalling=fxs_ks context=outbound channel=>1-4 Your contribution will be highly appreciated. Kind Regards, Mahmoud __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005...

...@belcentrale-incoming/622 .... register => 31307115627:secret@belcentrale-incoming/627 each user registers with something like this: [siemens1](xanadu-internal) type=friend callerid=Theo Band context=xanadu-thba [belcentrale-out-thba](belcentrale-outgoing) type=peer fromuser=31307115622 My extenson.conf looks like this: [xanadu-thba] exten => _+.,1,goto(00${EXTEN:1},1);00 is long distance calls => + exten => _0[1-9].,1,goto(0031${EXTEN:1},1);local calls =>0031 exten => _0031Z., 1,Macro(dialout,SIP/${EXTEN}@belcentrale-out-thba) ;NL exten => _+ZXX., 1,Macro(dialout,SIP/00...

...risk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it, my phone starts ringing that extension instead of accepting the call. I can only acept the call from other sides if I enter *8 (send) button. Here I am sending an example from my sip.conf file (I have sip accounts from 60-90) : [62] ; telefon Callgroup=3 pickupgroup=1-20 subsc...

This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I

Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. My goal is to run an asterisk box at each end with a t1 card and Ethernet card to