search for: _91nxxnxxxxxx

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition? Thanks, Erick. --

TRUNKMSD1=1 ; MSD digits to strip (usually 1 or 0) TRUNKMSD2=2 ; MSD digits to strip (usually 1 or 0) ; logn distance calls exten => _91NXXNXXXXXX,1,NoOp("Dialing: "${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,3,Congestion When I dial a long distance number (916503270309 for example) I get the message (I think from SBC) saying I must first dial a 1. Oth...

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that doesn't work. Has anybody tried this before?

...have in sip.conf [101] type=friend context=sipphones call-limit=99 callerid="iuser 101" disallow=all allow=ulaw allow=alaw username=101 secret=Passwd dtmfmode=rfc2833 host=dynamic mailbox=101 at default nat=yes canreinvite=no this is what i have in extensions.conf [outbound] exten => _91NXXNXXXXXX,1,Set(CALLERID(num)=mycompanyinc) exten => _91NXXNXXXXXX,2,Dial(SIP/att/${EXTEN:1},80) exten => _9NXXXXXX,1,Set(CALLERID(num)=mycompanyinc) exten => _9NXXXXXX,2,Dial(SIP/att/${EXTEN:1},80) any ideas? as this happens random, -------------- next part -------------- An HTML attachment was sc...

Does anyone have a Sipura 3k running, and using the FXO? I've got things working right, but if I try to toss a *67 in the dialplan, it seems the sipura is throwing a 403 forbidden back. For example: exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/${EXTEN:1} works fine exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/*67${EXTEN:1} does not (even if I toss a couple Ws in) I can't seem to get it to work in Sipura's internal dialplan either. Any suggestions? Maybe I'm missing something really, really st...

...face TRUNK=Zap/3 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [trunkint] exten => _9011.,1,Macro(dundi-e164,${EXTEN:4}) exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [trunkld] exten => _91NXXNXXXXXX,1,Verbose("LD") exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW) exten => _91NXXNXXXXXX,n,Hangup() [trunklocal] exten => _9NXXXXXX,1,Verbose("LOCAL") exten => _9NXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW) exten => _9NXXXXXX,n,Hangup() [trun...

...bridged=no echotraining=yes echotraining=600 rxgain=10.0 txgain=6.0 group=1 callgroup=1 pickupgroup=1 immediate=no amaflags=default accountcode=t1external busydetect=yes busycount=3 musiconhold=default jitterbuffers=4 channel => 1-12 /etc/asterisk/extensions.conf (with workaround) exten => _91NXXNXXXXXX,1,Answer(150) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91NXXNXXXXXX,3,Congestion Is there any mistakes in configuration or proper solutions to this problem? thank you. --

Our telco has setup toll access account codes for outgoing calls. I would like to include these account codes in the dialplan for certain extensions (fax lines, modems) so that they are not prompted for the 4 digit code when making a toll call. I have played around with the 'w' command with ZAP channels, commas, and DISA, all with no success. I see a lot of examples of Asterisk with

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

Anyone know of any GUI's that can be used to manage/setup asterisk?

...dor) and they have said the screening table has been disabled, so anything my phone system sends for callerid should be passing along. However, when I make a call, the primary number assigned on our PRI is showing up, not the value I've set. Here is how I'm making the call... exten => _91NXXNXXXXXX,1,Set(CALLERID(all)="Corner Homecare" <12703653903>) exten => _91NXXNXXXXXX,2,Dial(DAHDI/G2/${EXTEN:1}}) I'll be glad to offer up any debugging information that you might need, if it will help. I'm using Asterisk 1.6.0.14 and DAHDI 2.2.0. TIA -jonathan

...,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}) exten => _91NXXNXXXXXX,n,Congestion() exten => _91NXXNXXXXXX,n,Hangup() [hang-up] ; Hang up ; exten => s,1,Playback(thank-you-for-calling) exten => s,n,Playback(goodbye) exten => s,n,Hangup()...

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

...ocess of writing a PHP interface for * conf files. we are parsing the files like INI files but the only prob I have so far is that separate extensions in a context dont have any unique tag that I can capture. This works ok [trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion [international] ; ; Master context for international long distance ; ignorepat => 9 include => longdistance include => trunkint This does not, how do I separate the extensions apart from the [default] context so that PH...

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

...7) exten => 201,1,Macro(stdexten,SIP/201) [voicemail] exten => 300,1,Answer exten => 300,2,VoicemailMain(ssn-voicemail-greeting) exten => 300,3,Hangup [local] exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion [longdistance] exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion [macro-stdexten] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(default,s,1) exten => s-BUSY,1,Voicemail(b${MACRO...

...e sent to: [outbound-fax] exten => 8447,1,Answer() exten => 8447,n,GoSub(Capture-Fax,s,1) exten =>_NXXNXXXXXX,1,Answer() exten =>_NXXNXXXXXX,n,GoSub(DialOut-PSTN,s,1(1${EXTEN})) exten =>_1NXXNXXXXXX,1,Answer() exten =>_1NXXNXXXXXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN})) exten =>_91NXXNXXXXXX,1,Answer() exten =>_91NXXNXXXXXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN:1})) Actual outbound faxes work correctly. That is, a call from 173 to an outside fax machine works. The test faxes go to: [Capture-Fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRF...

...the following fragment which was designed to dial toll free on the PSTN and all other long distance on VoIP: [longdistance] include => local ;Handle local, etc first. (or so I thought!) exten => _91NXXNXXXXXX,1,Dial(${VPLSTRUNK}/${EXTEN:1}) ;Dial long distance through VoiP exten => _91NXXNXXXXXX,2,Congestion ;OOPS! No lines available? : : [local] : exten => _91800NXXXXXX,1,Dial(${PSTNTRUNK}/${EXTEN}) ; Long distance toll free accessed...

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion