search for: _011

...sions.conf, the first three patterns are for toll-free numbers, and fourth pattern is for other numbers, where an AGI is called for authentication. now when I dial 011448000664327 if falls into the fourth pattern, where as it should be matched by the first pattern. Any suggestions 1 - exten => _01144800XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 2 - exten => _01144808XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 3 - exten => _01144500XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 4 - exten => _011.,1,AGI(iax.agi) 4 - exten => _011.,2,Dial(${MAG}/${EXTEN:3},45,tT) 4 - exten => _011.,103,playbac...

Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten => _011870.,1,Goto(intl-disabled,s,1) exten => _01137455.,2,Goto(intl-disabled,s,1) exten => _01137477.,3,Goto(intl-disabled,s,1) exten => _0113749.,4,Goto(intl-disabled,s,1) exten => _011.,5,Goto(intl-disabled,s,1) exten => _011.,6,Playback(all-outgoing-lines-unavailable) exten => _011.,...

Hi, okay heres what I want to do .. simple ivr, we take a call, answer it, play a menu, dial out based on options. No problems so far. The CDR always shows the call as answered as I answer the 1st leg to play the prompts, I am actually more interested in if the 2nd leg - the outbound part - has been answered or not before the call is hungup. How can I get this and record the information in

...ID(name)="Stephen Reese") exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound) exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@vitel-outbound) exten => _011.,1,Set(CALLERID(num)=9045622082) exten => _011.,n,Set(CALLERID(name)="Stephen Reese") exten => _011.,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _911,1,Set(CALLERID(num)=9045622082) exten => _911,n,Set(CALLERID(name)="Stephen Reese") exten => _911,n,Dial(SIP/911...

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created

...}) >exten => _NXXXXXX,102,Dial(${NUFONE}/1${AREACODE}${EXTEN}) >exten => _NXXXXXX,203,Congestion() > >[iaxprovider] >exten => _1NXXNXXXXXX,1,Dial(${NUFONE}/${EXTEN}) >exten => _1NXXNXXXXXX,102,Dial(${TRUNK}) >exten => _1NXXNXXXXXX,203,Congestion() >exten => _011.,1,Dial(${NUFONE}/${EXTEN}) >exten => _011.,102,Congestion() >exten => _1011.,1,Dial(${NUFONE}/${EXTEN}) >exten => _1011.,102,Congestion() > >--Ernest

Hello all, Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies. So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10}) to determine which number is being dialed by the caller and then using a gotoif to get to

...ID(name)="Stephen Reese") exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound) exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@vitel-outbound) exten => _011.,1,Set(CALLERID(num)=9045622082) exten => _011.,n,Set(CALLERID(name)="Stephen Reese") exten => _011.,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _911,1,Set(CALLERID(num)=9045622082) exten => _911,n,Set(CALLERID(name)="Stephen Reese") exten => _911,n,Dial(SIP/911...

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a

...gt; voipms-inbound > include => voipms-outbound > > [voipms-outbound] > exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@voipms) > exten => _1NXXNXXXXXX,n,Hangup() > exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms) > exten => _NXXNXXXXXX,n,Hangup() > exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms) > exten => _011.,n,Hangup() > exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms) > exten => _00.,n,Hangup() > > ; inbound context example for your DID numbers, do not add the number > 1 in front > > [voipms-inbound] > exten => {redacte...

...(OH323_OUTCODEC=ulaw) exten => _NXXNXXXXXX,2,Dial(OH323/001${EXTEN},60,r) ; grab any 1NXXXNXXXXXX and add a 00 for h323 gatekeeper exten => _1NXXNXXXXXX,1,SetVar(OH323_OUTCODEC=ulaw) exten => _1NXXNXXXXXX,2,Dial(OH323/00${EXTEN},60,r) ; international calls via h323 gatekeeper exten => _011.,1,SetVar(OH323_OUTCODEC=ulaw) exten => _011.,2,Dial(OH323/00${EXTEN}) Here is the oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=no inBandDTMF=yes silenceSuppress...

...have been trying to setup a Voxee Sip termination. If anyone has extensions.conf different than Voxee suggestion. Can you please send me a copy? Thanks! Jerry Voxee web site advises to use: [voxee] exten => _1NXXNXXXXXX,1,Dial,SIP/${EXTEN}voxee exten => _1NXXNXXXXXX,2,Hangup exten => _011.,1,Dial,SIP/${EXTEN}voxee exten => _011.,2,Hangup register=>userid:password@66.246.246.52 sip.conf Settings [voxee] type=friend username=userid secret=password host=66.246.246.52 fromuser=userid dtmfmode=rfc2833 *************************************************************************...

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks.

I am trying to allow users to place outgoing international calls from a GPX-2000 with "early dial" enabled, connected to Asterisk 1.2.12.1 I have the following extension line: exten => _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) When I attempt to place a call to a number in, for instance, Kenya, I dial "011254"...etc. and I get this on the asterisk console: Executing Dial("SIP/1001-081fb718", "Zap/g1/0112") in new stack -- Called g1/0112 It is attem...

...untries how do I filter out cell phone calls ? Are calls to canadian mobile phones also unlimited ? Does anyone know ? Is the anywhere a List of all cellphone prefixes for that 19 countries ? I did not found something with google. That is what I found in the broadvoice wiki : -------- exten=_01130.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30) exten=_01131.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30) exten=_01132.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30) exten=_01133.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30) exten=_01134.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30) exten=_011351.,1,dial(SIP...

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ox=23@default host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="SIP Lite" <23> [sip-out] type=peer host=209.XXX.XXX.113 -----------------Extensions_additional-------------------------- [outrt-001-sip-out] include => outrt-001-Prizm-custom exten => _011.,1,Macro(dialout-trunk,4,${EXTEN},) exten => _011.,2,Macro(dialout-trunk,1,${EXTEN},) exten => _011.,3,Macro(outisbusy) ; No available circuits exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,4,${EXTEN},) exten => _1NXXNXXXXXX,2,Macro(dialout-trunk,1,${EXTEN},) exten => _1NXXNXXXXXX,3,Ma...

I want to do something like: exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _91NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9011.,1,Dial(Dial({TOLL}/${EXTEN}) (adapted from the book) but don't know where to put those lines....

...Finally, the number just before @voipjet in the Dial string is your VoipJet userid # exten => _1NXXNXXXXXX,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ exten => _1NXXNXXXXXX,2,Dial,IAX2/9333@voipjet/${EXTEN} ; VoipJet.com NANPA exten => _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ. exten => _011.,2,Dial,IAX2/9333@voipjet/${EXTEN} ; VoipJet.com WORLD ;Do not change IAX2/9333 in the above two lines! Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound clarity...