search for: _9nxxnxxxxxx

...les: ***/etc/zaptel.conf*** # Zaptel conf fxoks=1 fxsks=3-4 loadzone=us defaultzone=us ***/etc/asterisk/extensions.conf*** [globals] RECEPTIONIST=Zap/1 LOCALTRUNK=Zap/4 [incoming] exten => s,1,Answer() exten => s,2,Dial($(RECEPTIONIST)) [internal] [outgoing] ignorepat => 9 exten =>_9NXXNXXXXXX,1,Dial(${LOCALTRUNK}/${EXTEN:1}) exten =>_9NXXNXXXXXX,2,Playback(invalid) exten => _9NXXNXXXXXX,3,Hangup ***/etc/asterisk/zapata.conf*** language=en context=default switchtype=national signalling=fxo_ks channel => 1 signalling=fxs_ks channel => 3 channel => 4 Thats my entire setu...

...xxxxxxxxx secret=pppppppppp username=xxxxxxxxxx insecure=very context=from-broadvoice authname=xxxxxxxxxx dtmfmode=inband dtmf=inband authuser=xxxxxxxxxx ;Disable canreinvite if you are behind a NAT canreinvite=no quality=yes === Extensions.conf =========== ; I only addedd: [VOIP-OUT] exten => _9NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) exten => _9NXXNXXXXXX, 2, congestion() exten => _9NXXNXXXXXX, 102, busy() -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050308/4b1b1e3d/attachment.h...

Hi All, I have the following scenario, it may already have been answered elsewhere, but I cant find the solution. I already have a PBX and would like to start implementing asterisk. I have ordered a 4 port card from the asterisk store (2 port FXS and 2 port FXO) and am waiting for it to arrive. I do not want to plug my incoming lines into my FXO ports yet as not all the desks have IP phones

...*** ; BEGIN - Outbound Call Handling ;******************************************************************** ; [outbound-local] exten => _9NXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1}) exten => _9NXXXXXX,n,Congestion() exten => _9NXXXXXX,n,Hangup() exten => _9NXXNXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1}) exten => _9NXXNXXXXXX,n,Congestion() exten => _9NXXNXXXXXX,n,Hangup() exten => 911,1,Dial(${TRUNK0}/911) exten => 9911,1,Dial(${TRUNK0}/911) [outbound-long-distance] exten => _91NXXNXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1})...

...LERIDNUM}) [from-telx-nyc] exten => _7XXX,1,Dial(SIP/7101,20) exten => _7XXX,2,Voicemail(u7101) exten => _7XXX,102,Voicemail(b7101) exten => _7XXX,103,Hangup [macro-telx-nyc] exten => s,1,Noop() exten => s,2,Dial(IAX2/telx-nyc/${ARG1}) [outgoing] ;ingnorepat => 9 exten => _9NXXNXXXXXX,1,Noop() exten => _9NXXNXXXXXX,2,Macro(telx-nyc,${EXTEN}) exten => _9NXXNXXXXXX,3,Playback(invalid) exten => _9NXXNXXXXXX.4,Hangup ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 4. iax.conf (telx-nys17s) cat iax.conf [general] allow=all jitterbuffer=no tos=lowde...

I got a call from GV on Monday evening telling me they wanted me to move my Asterisk server over to a new IP address (216.229.127.40) by this saturday. Why the couldn't tell me this in an email is beyond me but anyways .. So I done changed the number and so far its all ok but whilst testing I noticed that I could no longer accept incoming phone calls. I swapped back and still no inbound

...on exten => 600,2,Echo ; Do the echo test exten => 600,3,Playback(demo-echodone) ; Let them know it's over exten => 600,4,Goto(s,6) ; Start over exten => 8500,1,VoicemailMain exten => 8500,2,Goto(s,6) [default] include => demo ; I modified stuff from here down... exten=_9NXXNXXXXXX, 1, dial(SIP/${EXTEN}@broadvoice-out,30) exten=_9NXXNXXXXXX, 2, dial(SIP/${EXTEN}@broadvoice-out2,30) exten=_9NXXNXXXXXX, 3, congestion() exten=_9NXXNXXXXXX, 103, busy() [sip] exten => 1,1,Dial(SIP/xlite1,20,tr) exten => 2,1,Dial(SIP/kphone1,20,tr) exten => 3,1,Dial(SIP/xlite2,20,tr) exte...

Is there an easy way to get information about local calling areas out of telcos? I'm trying to get a list of area codes and prefixes in my local calling area out of Verizon, and it looks like they've stopped providing the information online. Is there an easy source that I'm missing, or do I need to call them and have them mail me a copy every few months? Scott

...9 exten => _9011.,1,Dial(Zap/g2/${EXTEN:1}) exten => _9011.,2,Congestion include => longdistance [longdistance] ignorepat => 9 exten => _91NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion include => local [local] ignorepat => 9 exten => _9NXXNXXXXXX,1,Dial,MGCP/aaln/3@192.168.0.241 exten => _9NXXNXXXXXX,2,Congestion [outbound-default] include = extensions include = directdial include = longdistance include = local IPD:/etc/asterisk# **************************************************************************************...

...*************************************** > > ; > > [outbound-local] > > exten =3D> _9NXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1}) > > exten =3D> _9NXXXXXX,n,Congestion() > > exten =3D> _9NXXXXXX,n,Hangup() > > > > exten =3D> _9NXXNXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1}) > > exten =3D> _9NXXNXXXXXX,n,Congestion() > > exten =3D> _9NXXNXXXXXX,n,Hangup() > > > > exten =3D> 911,1,Dial(${TRUNK0}/911) > > exten =3D> 9911,1,Dial(${TRUNK0}/911) > > > > [outbound-long-dist...

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A

...ease 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer host = sip.net2phone.com username = PHONENUMBER secret = PASSWORD fromuser = PHONENUMBER fromdomain = net2phone.com context = incoming insecure = very canreinvite = no *extensions.conf* [outgoing] exten => _9NXXNXXXXXX,1,Dial(SIP/net2phone/${EXTEN:1}) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060509/99a8f265/attachment.htm

Started a new thread as my problem is somewhat different than the OP. Seems his somewhat different problem doesn't work as advertised either. Eric Wieling wrote: > I don't know what the problem is, but this is what I use and it works on my analog FXO port. > exten => _9NXXNXXXXXX,1,Dial(${PSTN}/w${EXTEN:1}) So, I modified slightly to fit my dialplan: exten => _8NXXXXXX,1,Dial(ZAP/g1/w${EXTEN:1}) and it MAY wait, but NOT off hook. The problem still remains, Asterisk begins to dial before dial tone is returned from the CO This was determined by monitoring the CO line w...

I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() { ... } and called in extensions.ael with: &foo(...) These are not deprecated because they use Gosub() underneath and not Macro()? Still good...

...repat => 9 exten => _9011.,1,Dial(Zap/g2/${EXTEN:1}) exten => _9011.,2,Congestion include => longdistance [longdistance] ignorepat => 9 exten => _91NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion include => local [local] ignorepat => 9 exten => _9NXXNXXXXXX,1,Dial,MGCP/aaln/3@192.168.0.241 exten => _9NXXNXXXXXX,2,Congestion [outbound-default] include = extensions include = directdial include = longdistance include = local and the *** mgcp.conf**** IPD:/etc/asterisk# more mgcp.conf ; MGCP Configuration for Asterisk [general] port=2727 b...

...K}) exten => _1NXXNXXXXXX,1,Set(CALLERID(all)="Viagenie (418-555-5555)") exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@${SIPTRUNK}) exten => _9NXXXXXX,1,Set(CALLERID(all)="Viagenie (418-555-5555)") exten => _9NXXXXXX,2,Dial(SIP/418${EXTEN:1}@${SIPTRUNK}) exten => _9NXXNXXXXXX,1,Set(CALLERID(all)="Viagenie (418-555-5555)") exten => _9NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@${SIPTRUNK}) exten => _91NXXNXXXXXX,1,Set(CALLERID(all)="Viagenie (418-555-5555)") exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:2}@${SIPTRUNK}) exten => _9.,1,Set(CALLERID(all)=...

...es, including outbound extensions are in this context exten => beginmenu,1,SetVar(CALLEDNAME=CompanyName) exten => beginmenu,2,Wait,1 exten => beginmenu,3,Background(company-main) exten => beginmenu,4,Background(ifyouknow) exten => beginmenu,5,Goto(company_mainmenu,s,1) exten => _9NXXNXXXXXX,1,Dial(IAX2/company-asterisk@outbound-asterisk/${EXTEN:1}) exten => 502,1,Dial(SIP/whatever1&SIP/whatever2|30|m) ... [company_mainmenu] exten => s,1,Background(company-nav1) exten => 1,1,Goto(company_sales,s,1) ; Sales exten => 2,1,Goto(companyname,502,1) ; Accounting exten => 3...

Can anyone send default configuration files to me for connect to broadvoice. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk Users, I have never done a dial plan for this scenario before. Is it possible to have Sip Phones make outbound calls through the PSTN? What would the call routing/dial plan would look like? -John _________________________________________________________________ Messenger Caf? ? open for fun 24/7. Hot games, cool activities served daily. Visit now.

This is not an Asterisk specific question but doesn't anyone know if you can automatically prepend a 9 on the call lists so clients can return dial without having to repunch in the number? If you go to directories now it just shows the number without a 9 (obviously). Maybe on the Asterisk side?? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: