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Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

..._1855.,1,Dial(${DIALOUTANALOG}/{${EXTEN}) exten => _1855.,2,Macro(fastbusy) exten => _1855.,102,Macro(dialoutsip,${EXTEN}) exten => _1800.,1,Dial(${DIALOUTANALOG}/${EXTEN}) exten => _1800.,2,Macro(fastbusy) exten => _1800.,102,Macro(dialoutsip,${EXTEN}) [long-distance] exten => _1XXXXXXXXXX,1,Macro(dialoutsip,${EXTEN}) exten => _1XXXXXXXXXX,2,Macro(fastbusy) exten => _1XXXXXXXXXX,102,Dial(${DIALOUTANALOG}/${EXTEN}) exten => _1XXXXXXXXXX,103,Macro(fastbusy) [home] include => operator include => e911 include => forced-analog include => fwd-out include => informa...

...n. I did that thusly: exten => 101,1,Set(CALLERID(number)=1112223333) exten => 101,2,Set(CALLERID(name)=fiznucked) exten => 101,3,Dial(SIP/officeata1,20,tr) that works. But the callerid doesn't work when I try to call out through teliax. exten => _1XXXXXXXXXX,1,Set(CALLERID(number)=1112223333) exten => _1XXXXXXXXXX,2,DIAL(IAX2//username/@teliax/${EXTEN},30,tr) I've been calling my cell phone from one of the SIP phones, and I do receive the call on the cell, but all I see for the caller id is "unknown". Teliax tells me th...

Hello All !!! I have 4 PSTN lines in the PBX server 1,2,3,4. First line will be used by only one extension (i.e. for the boss) for incoming and outgoing. This line is dedicated for him only.( The remaining lines will be shared by the employees 1) Group A have access to lines 2 , 3 & 4. 2) Group B have access to lines 3 & 4 3) Group C have access to line 4

...le here. 3. I'd like all 1 XXX XXX XXXX numbers to go thru after the 9, unmolested. 4. After the 9, I'd like all other 7 digit numbers to be dialed 1307NXXXXXX. Simple enough, right? I wrote up the following: [outwork] ignorepat => 9 exten => _91XXXXXXXXXX,1,StripMSD,1 exten => _1XXXXXXXXXX,2,Dial,Zap/2/BYEXTENSION exten => _9754XXXX,1,StripMSD,1 exten => _754XXXX,2,Dial,Zap/2/BYEXTENSION exten =>_9NXXXXXX,1,StripMSD,1 exten =>_NXXXXXX,2,Prefix,1307 exten => _1307NXXXXXX,3,Dial,Zap/2/BYEXTENSION [workext] ;; the active context for an internal extension include => o...

...group=3 echocancel=yes signalling=fxo_ls channel => 49-72 immediate=yes Extensions.conf: [from-cb] exten => s,1,DISA,no-password|internal [internal] include => sip-stations include => iax-trunks include => outbound [outbound] exten => _1XXXXXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) exten => _XXXXXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) exten => _XXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) When I pickup a line, and hit any key I get: -- Starting simple switch on 'DAHDI/49-1' -- Hungup 'DAHDI/49-1' -- Starting simple switch...

...,Goto,outtrunk|17005554223|1 ; if they press 0-9 > exten=_*,1,Goto,outtrunk|17005554223|1 ; if they press * > exten=_#,1,Goto,outtrunk|17005554223|1 ; if they press # I just loaded the latest from CVS and reduced it down to a simple: exten=s,1,Goto,outtrunk|17005554223|1 [outtrunk] exten=_1XXXXXXXXXX,1,StripMSD,1 exten=_XXXXXXXXXX,2,Dial,Zap/g1/${EXTEN} and it still stops passing audio after many calls. Well, existing calls still pass audio, but new calls complete but neither party can hear the other. Since this is no more complicated than many asterisk installations, I wonder if we have ba...

<<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>> Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf

For years we've had the following simple context for outgoing calls: [outtrunk] ; match any NANP, and strip leading 1 off exten => _1XXXXXXXXXX,1,StripMSD,1 ; dial outbound on trunk group 1 exten => _XXXXXXXXXX,2,Dial,Zap/g1/${EXTEN} But when I upgraded on Friday to the latest CVSHEAD, this no longer works. If I send 13115552368 to this context, I get a message like pbx.c: Channel 'Zap/361-1' sent into invalid extension ...

I have 2 X100P card and configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt I changed the area codes to match mine. When I try to dial out I get app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' A zap show channels gives me this Chan Extension Context Language MusicOnHold 1

...er id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default exten=_1XXXXXXXXXX!,1,Macro(trunkdial,${<my_trunk>}/${EXTEN:0},${trunk_cid}) comment=_1XXXXXXXXXX!,1,<my_trunk>,standard exten=_*811XXXXXXXXXX!,1,Macro(private_trunkdial,${<my_trunk>}/${EXTEN:3},null) comment=_*811XXXXXXXXXX!,1,<my_trunk>,standard (I replaced the name of my termination wih &l...

...; 300,1,Macro(stdexten,1234,SIP/300) exten => 301,1,Macro(stdexten,1234,SIP/301) exten => s,1,Dial(SIP/300,10) exten => s,2,Dial(SIP/301,10) exten => s,3,Dial,ZAP/2/XXXXXXX,10 exten => s,4,Voicemail,1234 exten => s,5,Hangup exten => _XXXXXXX,1,Dial,ZAP/1/${EXTEN} exten => _1XXXXXXXXXX,1,Dial,ZAP/2/${EXTEN} Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please contact support@ke6upi.com if you have questions about this email.

If this handles the case where 10 digits are required: exten => _9NXXXXXXXXX,1,StripMSD,1 exten => _NXXXXXXXXX,2,Dial,Zap/4/BYEXTENSION How do you create a handler which works for either this or the case with a leading '1' plus 10 digits? tnx -kim -- w8hdkim@gmail.com

I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that

How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet Example in my extension.conf I have: [iaxtel] exten => _1700NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1888NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten =>

...ile. Leave them here. ; ;--------------------------------------------------------------------- ;============================== GLOBALS ============================== ;--------------------------------------------------------------------- ; [globals] ; ;Dial Patterns LOCAL_PATTERN=_NXXXXXX LD_PATTERN=_1XXXXXXXXXX INT_PATTERN=_011. FORCE_SIP_LOCAL_PATTERN=_8NXXXXXX FORCE_SIP_LD_PATTERN=_81XXXXXXXXXX FORCE_SIP_INT_PATTERN=_8011. FORCE_FXO_LOCAL_PATTERN=_9NXXXXXX FORCE_FXO_LD_PATTERN=_91XXXXXXXXXX FORCE_FXO_INT_PATTERN=_9011. ; ;--------------------------------------------------------------------- ;=========...

...> ; Outbound > > [fwd-out] > exten => _8.,1,SetCallerID(${FWDUSERID}) > exten => _8.,2,SetCIDName(${FWDUSERNAME}) > exten => _8.,3,Dial(SIP/${EXTEN:1}@fwd,70) > exten => _8.,4,Macro(fastbusy) > exten => _8.,5,Hangup > > [long-distance] > exten => _1XXXXXXXXXX,1,Macro(dialoutsip,${EXTEN}) > exten => _1XXXXXXXXXX,2,Macro(fastbusy) > > ; SIP lines > > [inbound-sip] > exten => 533990,1,Goto(local,101,1) > > ; Internal Extensions > > [local] > exten => 101,1,Dial(${MAINPHONE},20,Tt) > exten => 101,2,Voicema...

Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its nothing special but it may be of use to the newbies.. Like me.. If you would like a copy let me know and I will send it to you.. later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze