search for: _1x

...s g723.1 pass-thru! allow=g729 ; Pass-thru only unless g729 license obtained ****** extensions.conf ****** [phones] include=>internal include=>remote [internal] exten=>_2x,1,NoOp() exten=>_2x,n,Dial(SIP/${EXTEN},30) exten=>_2x,n,Hangup() [remote] ;exten=>_1x,1,NoOp() exten=>_1x,1,Dial(SIP/asterisk2/${EXTEN}) exten=>_3x,1,Dial(SIP/asterisk2/${EXTEN}) exten=>_1x,n+101,Hangup() exten=>_3x,n+101,Hangup() [asterisk2_incoming] include=>internal ************************************************** asterisk 2 **********************************...

...asic extension.conf I set up a pattern starting with "1" and a second digit should dial the sip extension entered by the user and if the user don't pick up or is unavailable the call goes to the user voicemail and then hangup. This basic setup can be seen next: [default] exten => _1X,1,Dial(SIP/${EXTEN},10) exten => _1X,2,VoiceMail(${EXTEN}@default,u) exten => _1X,3,HangUp() Now, what happens if the user dials 15? Then the pattern is applied and the asterisk tries to dial that sip extension that doesn't exist, the next step that is the voicemail also fails as 15 is n...

First, here's what I want to do / what I have: X100P and a Quicknet PhoneJack. I want to be able to pick up the analog phone (connected to the phonejack) and dial another computer (with the same hardware) or just make a regular phone call which will be decided by asterisk depending on the phone number dialed. I know that this won't be taking full advantage of asterisk, but I'm

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at IAX)},i)}>10]?fail) exten => continue,n,Goto(from-pr...

...er, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten => _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ulaw ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes ins...

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R

...ation of winnowing down the options to the most basic config. The same phone works flawlessly with other lines that are configured on it to other * servers. Here is my entire relevant configuration. It's as simple as you can get, really. I dial 14109850123 (as a test number - it matches the _1X. list) and I get an almost instant hangup. --------------- ;sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls dtmfmode=rfc2833 allow=all [3015321510] type=friend...

...ername=2 secret=1234 amaflags=documentation accountcode=12 subscribecontext=IntPhones callerid="phone 12" <12> limitonpeers=yes call-limit=100 ======================================= and on extensions.conf my dial lines are like: ======================================= exten => _1X,1,Dial(SIP/${EXTEN:1},,tTr) exten => _1X,n,Hangup() ======================================= Can anyone help me? I don't underwstand where I make the mistake! Thanks to everyone Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/...

...this seems mandatory for SIP. I find the help text misleading : "NOTE: Most channel types need to be in Answer state in order to receive DTMF". This is my test dialplan: [earlydial] ; Test Early Dial exten => 1,1,Verbose(2,Incomplete 1 test) same => n,Incomplete(n) exten => _1X,1,Verbose(2,Incomplete 1X test) same => n,Incomplete(n) exten => _1XX,1,Verbose(2, Dialed ${EXTEN}) same => n,Playback(extension) same => n,SayDigits(${EXTEN}) same => n,Hangup() It works, but seems a bit complicated: is this the correct way to use Incomplete...

...utbound [ContextB] exten => 20,1,Dial(SIP/20,20) exten => 21,1,Dial(SIP/21,20) exten => 22,1,Dial(SIP/22,20) include => outbound [ContextC] exten => 30,1,Dial(SIP/30,20) exten => 31,1,Dial(SIP/31,20) exten => 32,1,Dial(SIP/32,20) include => outbound [default] exten => _1X,1,GoTo(ContextA,${EXTEN},1) exten => _2X,1,GoTo(ContextA,${EXTEN},1) exten => _3X,1,GoTo(ContextA,${EXTEN},1) [outbound] exten => _9XXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@192.168.1.100) So each user registers and they can call each other and they can dial 9xxxxxxxxxx to dial local and ld. T...

...ems to work: [earlydial] ; Test Early Dial exten => _.,1,Set(l_Extension=${EXTEN}) exten => _.,n,Goto(earlydial2,${l_Extension},1) [earlydial2] exten => _.,n,Goto(noMatch,1) exten => noMatch,1, Incomplete(n) exten => i,1,Goto(noMatch,1) exten => t,1,Goto(noMatch,1) exten => _1XX,1,Verbose(2, Dialed ${EXTEN}) same => n,Playback(extension) same => n,SayDigits(${EXTEN}) same => n,Hangup() Best regards, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I've been using Grandstream phones for more than 10 years, but only yesterday tried to use Early Dial... and I failed. What is needed on the Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip on Asterisk-13.7.1. Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise

Scenario: Both boxes are behind firewall, port udp 4569 is open. If I don't want the username and password in dialing string do I have to use register statement in IAX.CONF. Can anybody post some working samples; I have a hard time making it to work with the samples posted on wiki. -- #Joseph

Dear, the goto function to the iax dialing, makes bill duration and call duration wrong, in cdr.they are equal to ringing time. the cdr will be produced and saved into the dbase, when the callee picks up the phone. is any way to have real duration time ? [main] exten => _1X.,1,GOTO(LOPL,${EXTEN},1) .... [LOPL] exten => _X.,1,Dial(IAX2/MAIN/${EXTEN},60)

Hi all, I'm trying to setup a simple IVR menu in a context in extensions.conf. So far, I have: extension s for playing back the menu # to repeat it * for directory 0 for operator 1 which goes to another context: exten => 1,1,GoTo(option_1,s,1) Here is what I have in extensions.conf: [incoming] ; main greeintg exten => s, 1, Ringing exten => s, 2, Wait(10) exten => s, 3, NoOp()

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx