search for: _nxxxxxxx

2015-02-26 10:45 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > > You just need to use call groups. > > In your chan_extra.conf (if it's an OpenVox) or chan_dahdi.conf, add > something like > group=1 > to the definition for each span. > > Now in the [globals] section of your dialplah, have something like > MOBILE=EXTRA/r1 > for an

...2-27 10:25 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > O.K. So what does your existing Dial() statement in extensions.conf look > like? > apology, put the gateway was sangoma but is a openvox , all my outgoing calls out for this context: [my-mobile-out] exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT) exten => _NXXXXXXX,n,Dial(SIP/1004/${EXTEN},55,rT) exten => _NXXXXXXX,n,Dial(SIP/1001/${EXTEN},55,rT) exten => _NXXXXXXX,n,Dial(SIP/1002/${EXTEN},55,rT) exten => _NXXXXXXX,n,Playback(all-circuits-busy-now) exten => _NXXXXXXX,n,Hangup() my main number...

...00,1,Dial(SIP/88888888,20,rt) exten => 900,1,Dial(SIP/99999999,20,rt) exten => _ZXXX,1,Answer exten => _ZXXX,2,Dial(Zap/g1/${EXTEN}) exten => _ZXXXX,1,Answer exten => _ZXXXX,2,Dial(Zap/g1/${EXTEN}) exten => _NXXXXX,1,Answer exten => _NXXXXX,2,Dial(Zap/g1/${EXTEN}) exten => _NXXXXXXX,1,Answer exten => _NXXXXXXX,2,Dial(Zap/g1/${EXTEN}) [home] exten => s,1,Playback(greetings) exten => 100,1,Answer exten => 100,2,MusicOnHold() exten => 100,4,Hangup exten => 111,1,Dial(CONSOLE/dsp) exten => 111,4,Hangup exten => 700,1,Dial(SIP/88888888,20,rt) exten =&g...

...00,1,Dial(SIP/88888888,20,rt) exten => 900,1,Dial(SIP/99999999,20,rt) exten => _ZXXX,1,Answer exten => _ZXXX,2,Dial(Zap/g1/${EXTEN}) exten => _ZXXXX,1,Answer exten => _ZXXXX,2,Dial(Zap/g1/${EXTEN}) exten => _NXXXXX,1,Answer exten => _NXXXXX,2,Dial(Zap/g1/${EXTEN}) exten => _NXXXXXXX,1,Answer exten => _NXXXXXXX,2,Dial(Zap/g1/${EXTEN}) [home] exten => s,1,Playback(greetings) exten => 100,1,Answer exten => 100,2,MusicOnHold() exten => 100,4,Hangup exten => 111,1,Dial(CONSOLE/dsp) exten => 111,4,Hangup exten => 700,1,Dial(SIP/88888888,20,rt) exten =&g...

...[globals] IAXINFO=guest ; IAXtel username/password OUTGOING => Zap/1 [from-sip] ignorepat => 9 exten => _X.,1,Dial(Zap/g1/${EXTEN},60) exten => _X.,2,Hangup ;exten => _NXXXXXXX,1,Dial(Zap/g1) [default] exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer ; Answer the line exten => s,3,Dial(SIP/sipphone)

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled number]@[voip IP]> Call...

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.