search for: _1xx

...port or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten => 6101,hint,SIP/101 exten => 6102,hint,SIP/102 exten => 6103,hint,SIP/103 exten => 6104,hint,SIP/104 exten => 6105,hint,SIP/105 exten => _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3) exten => _1XX,2,Dial(SIP/${EXTEN},20,Tt) exten => _1XX,3,VoiceMail(${EXTEN}@default,u) exten => _1XX,104,VoiceMail(${EXTEN}@default,b) sip.conf [general] allowsubscribe=yes ;subscribecontext = default notifyringing=no no...

...fter, cause if I execute flash again, the control goes back to Asterisk. exten => s,1,Answer exten => s,2,Wait,1 exten => s,3,Background(IVR) exten => s,4,DigitTimeout,4 exten => s,5,ResponseTimeout,4 exten => t,1,Goto(operadora,s,1) exten => i,1,Playback(invalid) exten => _1XX,1,Flash exten => _1XX,2,background(silence/1) exten => _1XX,3,SendDTMF(${EXTEN}) exten => _1XX,4,background(silence/1) exten => _1XX,5,Hangup Thank you

Hi everyone, How can I add the equivalent of: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve

...yes [vvfarm] type=friend username=colo secret=testpassword auth=plaintext host=64.194.211.170 context=iax-incoming peercontext=vvfarm-extensions qualify=yes trunk=yes Extensions.conf on the same server [iax-incoming] exten => _###,1,Dial(SIP/17${EXTEN}-1,20) [remote-extensions] exten => _1XXXX,1,Dial(SIP/17${EXTEN}-1,20) exten => _1XXXX,n,Voicemail(${EXTEN:0:3}|su) exten => _1XXXX,n,Dial(SIP/${EXTEN}-1) exten => _11XXX,1,Dial(iax2/vvfarm/${EXTEN:2}-1,20) Iax.conf on server B [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxd...

...(simzy1) ; input an extension exten => 301,4,WaitExten(8) exten => 301,5,Dial(SIP/100,15) ; goes to operator exten => 301,4,Wait(8) include => spa exten => _XXX,6,VoiceMail(100 at default) exten => 301,n,Hangup() [spa] exten =>_301,1,GoTo(sipura-line,${EXTEN},1) exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten => _1XX,2,VoiceMail(${EXTEN}@default) ; direct 2 voicemail box if line is busy or unavailable exten => _1XX,3,HangUp() exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3...

...exten => s,405,Dial(sip/${ARG1}5,45,r) exten => s,406,Goto(s-u,1) exten => s,506,Dial(sip/${ARG1}6,45,r) exten => s,507,Goto(s-u,1) exten => s-b,1,Voicemail(b${ARG1}) exten => s-u,1,Voicemail(u${ARG1}) when I want to ring mutiple phones this is what I do: [rollover] exten => _1XX,1,Dial(SIP/${EXTEN}1,45,r) exten => _1XX,102,Dial(SIP/${EXTEN}2,45,r) exten => _1XX,203,Dial(SIP/${EXTEN}3,45,r) exten => _1XX,304,Dial(SIP/${EXTEN}4,45,r) exten => _1XX,405,Dial(SIP/${EXTEN}5,45,r) exten => _1XX,506,Dial(SIP/${EXTEN}6,45,r) [default] exten => 160,1,Dial(Local/10...

Hello Everyone, I am using the following dialplan to allow users to check their messages from PSTN world: ; Internal Routing exten => _1XX,1,Dial(SIP/${EXTEN}, 10) exten => _1XX,n,Wait(1) exten => _1XX,n,Answer exten => _1XX,n,Wait(1) exten => _1XX,n,Voicemail(${EXTEN},us) exten => _1XX,n,Hangup The problem is that when the user presses `*#` to check his/her messages, it adds an additional message, even if there were n...

...t=n.n.n.n/255.255.255.0 ; n.n.n.n external NAT router address host=dynamic context=fullaccess ;allow=all In this way, I can call any 3XX extension using an IAX connection between the two PBX's. It works perfect. Now, on the other PBX I have the following in extensions.conf exten => _1XX,1,Dial(IAX/pspbx@n.n.n.n/${EXTEN}) exten => _1XX,2,Hangup exten => _1XX,102,Hangup where n.n.n.n is the external IP address of the NAT router. pspbx is the IAX user defined on the PBX behind the router, in iax.conf file like that: [pspbx] type=user username=pspbx auth=plaintext permit=0.0.0...

...fter, cause if I execute flash again, the control goes back to Asterisk. exten => s,1,Answer exten => s,2,Wait,1 exten => s,3,Background(IVR) exten => s,4,DigitTimeout,4 exten => s,5,ResponseTimeout,4 exten => t,1,Goto(operadora,s,1) exten => i,1,Playback(invalid) exten => _1XX,1,Flash exten => _1XX,2,background(silence/1) exten => _1XX,3,SendDTMF(${EXTEN}) exten => _1XX,4,background(silence/1) exten => _1XX,5,Hangup Thank you

...demo include => from-sip Configuration on pbx2: ---------------------- sip.conf: ========= [pbx1] type=friend username=pbx1 secret=pbx1 host=10.1.3.207 extensions.conf: ================ [from-sip] exten => 223,1, Dial(SIP/adamo2,10,t) exten => 258,1, Dial(SIP/okrasz2,10,t) exten => _1XX,1, Dial(SIP/pbx2:pbx2@10.1.3.207/${EXTEN},30,r) [default] include => demo include => from-sip ===================== end of configuration ==================== In this case when I start PBXes, register an okrasz client to pbx1 and write on pbx2 console: > dial 158 ------- I get fallowing...

...callgroup=1 pickupgroup=1 type=friend secret=sip2 auth=md5 host=dynamic reinvite=no canreinvite=no callgroup=1 pickupgroup=1 language=es dtmfmode=rfc2833 callerid="sip2" <102> context=telefonos extensions.conf: [globals] EXTEN106=Sip/sip1 EXTEN107=Sip/sip2 [telefonos] exten => _1XX,1,NoOp(${CALLERID}) exten => _1XX,2,Dial(${EXTEN${EXTEN}},,tT) exten => _1XX,3,Hangup exten => _1XX,103,VoiceMail2(u101) _________________________________________________________ Do You Yahoo!? Informaci?n de Estados Unidos y Am?rica Latina, en Yahoo! Noticias. Vis?tanos en http://no...

...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 ? Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en...

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

...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....

...h outgoing calls to VOIP, incoming that go to a Pana extensions work fine. --- Extensions.conf [dialstring] exten => t,1,Dial(Zap/g1/100,60,tn) exten => i,1,Dial(Zap/g1/100,60,tn) [from-e100p] include => dial-sip include => out-voip [dial-e100p] exten => _1XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)}) exten => _1XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID( num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN}) exten => _1XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0)) exten => _1XX,4,Dial(Zap/g1/${EXT...

Dear all I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ? Rgd satish patel

Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal ---------...

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

...ot; application and not continue (if in "Background", I can detect the tone). I even use the option M(macro) try to catch the tone send from Panasonic, but failed also. Anybody can give me your comment? Thanks! BR, Dominic ---------------------------------------- [via-net] exten => _1XX,1,Answer exten => _1XX,2,SetVar(called_ext=${EXTEN}) exten => _1XX,3,Goto(test,s,1) [test] exten => s,1,Background(transfer) exten => s,2,Dial(Zap/g1/${called_ext}|10|m) exten => s,3,NoOp ------------------------------------------ Message on console while FOREVER ringing, --...

...h outgoing calls to VOIP, incoming that go to a Pana extensions work fine. --- Extensions.conf [dialstring] exten => t,1,Dial(Zap/g1/100,60,tn) exten => i,1,Dial(Zap/g1/100,60,tn) [from-e100p] include => dial-sip include => out-voip [dial-e100p] exten => _1XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)}) exten => _1XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID( num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN}) exten => _1XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0)) exten => _1XX,4,Dial(Zap/g1/${EXT...