search for: _6xxx

Hi, I've noticed an unusual behavior on the dialplan execution: assume this DP: exten => _6XXX,1,NoOp(test1) exten => _XXXX,1,NoOp(test2) exten => _XXXX,2,NoOp(test3) If I call 6000 then test1 and test3 NoOps get executed, even though the pattern is different. I've always thought that if I call 6000 it would match the 6XXX pattern, that only has 1 priority, that would get execut...

...:password@66.151.89.145 [1000] type=user username=1000 auth=plaintext permit=0.0.0.0/0.0.0.0 host=dynamic context=fullaccess My extension.conf is as follows for the server that is behind NAT: [fullaccess] exten => _6XXX,1,Dial(IAX2/2000@66.151.89.145/${EXTEN}) exten => _6XXX,2,Hangup exten => _6XXX,102,Hangup --------------------------------------------- Free POP3 Email from www.Gawab.com Sign up NOW and get your account @gawab.com!!

...l seems to work ok, except that I get a load of "pbx.c: Timeout, but no rule 't' in context 'AgentQ'" in the error log. What would I use in the 't' rule to stop this error from ocurring ? /* extensions.conf portion for calling agent */ ... [AgentQ] exten => _6XXX,1,Dial(SIP/${EXTEN},12) ... /* extensions.conf portion for calling the q */ ... [macro-callq] exten => s,1,Answer() exten => s,2,GotoIfTime(${ARG4},${ARG5},*,*?s,4) exten => s,3,Goto(s,6) exten => s,4,Playback(${ARG3}) exten => s,5,Queue(${ARG2},nt,,,120) exten => s,6,Voicemail(s...

...-- Hungup 'Zap/25-1' I also see this error Mar 4 06:07:21 WARNING[13946]: chan_zap.c:7069 pri_fixup_principle: Call specified, but not found? Mar 4 06:07:21 WARNING[13946]: chan_zap.c:7711 pri_dchannel: Ringing requested on channel 0/1 not in use on span 1 Extensions.conf exten => _6XXX,1,Answer exten => _6XXX,2,Dial(ZAP/g1/${EXTEN}) exten => _6XXX,3,congestion() Zapata.conf [channels] switchtype=national context=from-pstn signalling=pri_cpe pridialplan=unknown usecallerid=asreceived echocancel=no echocancelwhenbridged=no echotraining=400 overlapdial=yes immediate=no grou...

...Centre) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(welcome-fxqueue) exten => s,n,Goto(5210,1) ; FxQueue exten => 5210,1,Noop() exten => 5210,n,Ringing() exten => 5210,n,Wait(2) exten => 5210,n,Queue(FxQueue|tH) exten => 5210,n,Hangup() exten => _6XXX,1,macro(ccexten,${EXT_${EXTEN}}) [macro-ccexten] exten => s,1,Set(EXT=${ARG1}) exten => s,2,GotoIf([$:{EXT}]?4:3) exten => s,3,Goto(i,1) exten => s,4,Dial(${EXT},10,tT) exten => i,1,Playback(pbx-invalid) -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322...

Heya Rodrigo Not sure, but this expansion on Sammy's concept may help you achieve the delayed ring on the secondary extensions you were looking for. exten => _600.,1,Dial(PJSIP/${EXTEN}) exten => _600.,n,Hangup exten => _600.wait5,1,Wait(5) exten => _600.wait5,n,Dial(PJSIP/${EXTEN:0:4}) exten => _600.wait5,n,Hangup exten => 555,1,Dial(LOCAL/6001&LOCAL/6002.wait5)

that was my first idea. and how should an other user know which number he should dial? user a: soft phone extension 100 hardware phone extension 101 On 06.02.19 15:25, Mitch Claborn wrote: > You can do this in the dial plan. Register the devices separately and > include both addresses in the Dial() command. > > > Mitch > > On 2/6/19 8:16 AM, basti wrote: >> In

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm

...ontext=default type=friend encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no disallow=all allow=ulaw dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass *extensions.conf:* [default] exten => _6XXX,1,Dial(SIP/${EXTEN}) *rtp.conf:* [general] rtpstart=10000 rtpend=20000 icesupport=yes stunaddr=stun.l.google.com:19302 2015-08-10 12:35 GMT-03:00 Joshua Colp <jcolp at digium.com>: > Marek Cervenka wrote: > >> hello, >> >> i'm facing strange problem >>...

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside