search for: _9xx

...cs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 Then in extensions.conf I have: [default] ; this below for internal extensions - works OK exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm) ; for dialing outbound - over ISDN line - this bit does not work exten => _9XX.,2,Dial(ZAP/g1/${EXTEN},60) exten => _9XX.,2,Hangup Error I get is: -- Executing Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stack Jul 25 11:56:33 NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/cong...

...roadvoice.com fromdomain=sip.broadvoice.com outboundproxy=proxy.nyc.broadvoice.com insecure=port disallow=all allow=gsm allow=g729 dtmfmode=rfc2833 dtmf=inband canreinvite=no context=from-broadvoice ntp_ip=ntp.broadvoice.com call-limt=1 Here is my configuration in the extensions.conf: exten => _9XX[2-5].,1,dial(SIP/01155${EXTEN:1}@sip.broadvoice.com,40) exten => _9XX[2-5].,2,congestion() exten => _9XX[2-5].,102,busy() How can I limit calls in my sip peers?? Everton Goularth Uberlandia - MG - Brazil _______________________________________________________ Yahoo! Mail - Sempre a m...

I have two Asterisk servers, one of which uses a switch statement (Server 2). On Server 2, the dialplan is as follows: [provider] switch... [default] include=>provider exten=>451,1,Dial,Zap/1 ... (No extensions defined for Server 2 are "can_match" (eg. exten=>_9XX...)) The problem is that when I pick up a phone and dial 451, it searches Server 1 before using the extension defined in the default context. Is there a way to set Asterisk up to search the local dialplan before checking the switched server? - Thank you for your time. __________________________...

I have SER running with Asterisk, both on seperate servers. If I call another SIP number from my SIP phone SER looks up the phone number to see if it's online. If it's not online it forwards the call to Asterisk. How do I configure the extensions.conf file so that calls being forwarded to Asterisk destined for VoiceMail do not conflict with normal outbound calls destined for the PSTN?

...should reside in Realtime. Realtime works just fine, so does my extensions in AEL. But not on the same time. This is how my extensions.ael look like now: context internal { 100 => Playback(tt-monkeys); 101 => Dial(SIP/cgm); 102 => Dial(SIP/bluecommand); _9XX => ?????????? 500 => Agi(agi://localhost/internal.agi); _8XXX => { NoOp("Calling ${EXTEN}"); Dial(SIP/${EXTEN}); } }; context database { eswitches { Realtime/ContextShouldBeHere@${C...

Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can

HI, I'm using netmeeting to connect to an asterisk server and dial out. my extension looks like this exten => s,1,Dial,Zap/1/ Unfortunatelly the number that I have dialed in Netmeeting is lost ;-( If I hardcode the number on the line above, like ... exten => s,1,Dial,Zap/1/6642794 ... everything works fine What am I missing?