search for: _xxx

...ks on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks??? [outbound] exten => _XXX.,n(dial),gotoif($[${loadbalance} = 1 ]?balance) ;at dial decide if we want to load balance exten => _XXX.,n,goto(50) ;if no balance then outbound call initiated with failover exten => _XXX.,n(balance),NoOp("Load Balancing Active") ;NoOp - display Load balancing status exten...

...dynamic option as all of our calls are database driven and we can't add a seperate entry per customer to the feautres.conf. [MSIP-DynPark] exten => s,1,NoOp(Dynamic Parking) exten => s,n,NoOp(Return Parked Call) exten => s,n,GoTo(${CUT(${l_ndeContext}-ndeArgs,~,1)},1) exten => _XXX,1,Set(PARKINGDYNAMIC=parkinglot_small) exten => _XXX,n,Set(PARKINGDYNEXTEN=110) exten => _XXX,n,Set(PARKINGDYNPOS=111-120) exten => _XXX,n,Set(PARKINGDYNCONTEXT=${account}-Lot) ;exten => _XXX,n,Set(PARKINGEXTEN=99) exten => _XXX,n,Park() [MSIP-DynParkPickup] exten => _NXX,1,Parke...

...n stopped working. Not sure what I changed. System Info: Asterisk 1.4.21.2 Queuemetrics 1.6.3.0 [queuedial] ; this piece of dialplan is just a calling hook into the [qm-queuedial] context that actually does the ; outbound dialing - replace as needed - just fill in the same variables. exten => _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3}) exten => _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3}) exten => _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)}) exten => _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER}) exten => _XXX.,n,Set(QueueName=${QDIALER_QUEUE}) *exten => _XXX.,n,MixMonitor...

...cal] exten => _XX,1,Set(DYNAMIC_FEATURES=nway-start) exten => _XX,2,SIPDtmfMode(inband) exten=> 10,3,Dial(SIP/saad,,tT) exten=> 10,n,Hangup exten=> 11,3,Dial(SIP/riz,,tT) exten=> 11,n,Hangup exten=> 12,3,Dial(SIP/rehmat,,tT) exten=> 12,n,Hangup [dynamic-nway] exten => _XXX,1,Answer exten => _XXX,n,Set(CONFNO=${EXTEN}) exten => _XXX,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite) exten => _XXX,n,Set(DYNAMIC_FEATURES=) exten => _XXX,n,MeetMe(${CONFNO},pdMX) exten => _XXX,n,Hangup [dynamic-nway-invite] exten => 0,1,Read(DEST,dial,,i) exten => 0,n,Se...

...some_sip_reg BUT, what I need to do is dynamically decide where the hint checks for buddy status, because I am using patterns in this context. In other words, I need to find out the values of ${some_sip_reg} before the using the hint priority. Ideally, something sort of like this: exten => _XXX,hint,Set(hint_reg=${EXTEN}-reg} exten => _XXX,hint,SIP/${hint_reg} exten => _XXX,SIP/${EXTEN}-reg} Or, even easier (if it can even be done) is write a function: exten => _XXX,hint,SIP/ReturnCorrectRegistration() What's the best way to approach my problem? Mike -------------...

...o make a system where a conference user can invite others to join. I am running the 1.2 version of asterisk, so can't use the example on voip-info.org. With use of the X flag on a meetme conference to exit with a single digit, I can get people to join me in a conference with exten => _XXX,1,Dial(${THEIR_EXTEN},,dG(conference-context^${CALLERID}^1)) where the conference-context has a something like this [conference-context] exten => _XXX,1,MeetMe(${EXTEN},XMsa) exten => _XXX,2,MeetMe(${EXTEN},Ms) The problem with that approach is that you never get to talk to the called pa...

Hello, i'm using the parking feature. When the call is parked by A (number *15) , B is correctly parked, by A did not hangup automatically. Here are the dialplan [local] exten => _XXX.,1,Wait(0) exten => _XXX.,n,Dial(SIP/${EXTEN:0}@trunk_sip_2,0,TK) exten => _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK) exten => _XXX.,n,Playback(callbox-thinkro-trunkDefautIndispo) exten => _XXX.,n,Hangup() exten => h,1,Hangup() and the logs : -- Executing [0383824377 at local:1] Wa...

Hi everybody, I have a problem using parking for outgoing call. A is an local sip phone. A is using the local extension : [local] exten => _XXX.,1,Wait(0) exten => _XXX.,n,Dial(SIP/${EXTEN:0}@trunk_sip_2,0,TK) exten => _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK) exten => _XXX.,n,Playback(callbox-thinkro-trunkDefautIndispo) exten => _XXX.,n,Hangup() exten => h,1,Hangup() When A is calling an outside number, trunk_sip_2 is used. Th...

Hello list, I want to get the phone number out of the following P-Asserted-Identity header : /"BlaBlaBla" <sip://88779922//@192.168.8.10;user=phone>"/ I do the following in the dialplan : /exten => _XXX.,n,Set(PY=${SIP_HEADER(P-Asserted-Identity)}) exten => _XXX.,n,Set(PY2=${CUT(PY,@,1)})/ This gives me : /"BlaBlaBla" <sip://88779922/ How can I extract /88779922/ out of this string ?? I'm trying this : /exten => _XXX.,n,Set(PY4=${CUT(PY2,\:,1-)}) / but this does not chan...

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

...567|30|W duration=9 billsec=0 uniqueid=1248584760.896 My dialplan rules look like this; I come in to cob-meetme from an extension earlier in the plan, with MY_ACCOUNTCODE and COB_CONFNO already set. The callout context places an outgoing call. [cob-meetme] exten => _XXX,1,Answer exten => _XXX,n,Set(CDR(accountcode)=${MY_ACCOUNTCODE}) exten => _XXX,n,Set(MEETME_EXIT_CONTEXT=cob-meetme-escape) exten => _XXX,n,MeetMe(${EXTEN},d1qMX) exten => _XXX,n,Hangup [cob-meetme-escape] exten => *,1,Set(CDR(accountcode)=${MY_ACCOUNTCOD...

...on rings? Here is what we have in extensions_additional.conf: exten => 100,1,Goto(ext-local,10100,1) exten => 101,1,Goto(ext-local,10101,1) exten => 102,1,Goto(ext-local,10102,1) exten => 103,1,Goto(ext-local,10103,1) Would something like this in extensions.conf work? exten => _XXX*,1,Voicemail(u${EXTEN:1}) Where would be the best place to put it? Thanks for any suggestions you can provide!

...sed. ? This works for a three digit extension but I want to send an any length extension that hits this context in entirety. If I use a? _x!? it just stops at the first character. Extensions I want to send to the far end are a combination of 2,3,4 and 10 digit numbers. [pbx_server] exten => _xxx,1,Answer exten => _xxx,n,Dial(IAX2/pbx/${EXTEN}) exten => _xxx,n,Playback(vm-nobodyavail) exten => _xxx,n,Hangup Doug -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131116/2cb4566d/attach...

Sorry that I wrote in german : Ich benutze asterisk mit dem zaphfc Treiber. Jetzt hab ich folgendes Problem, habe 2 ISDN-Telefone angeschlossen. zaphfc im nt-mode. Anrufe von ausserhalb per sip (sipgate.de) kommen an. Wenn ich aber intern zwischen den zwei Telefonen (Ascom Eurit 30) sprechen m?chte geht das nur wie folgt : Erst die Nebenstelle w?hlen und dann den H?rer am Telefon abnehmen.

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

'lo, all, Is there some (possibly undocumented?) way that I can pattern-match on a specified number of repeating digits? (Something similar to regular expressions' {}) Here's an example: let's say I have a string of things that need to be done for both extensions 233 and 255. I can either... A) Repeat the exact same code for both extensions, like so: exten =>

...ugh, channel 'SIP/200-006fa300' status is 'CHANUNAVAIL' I used sample I found the internet just to see if this is possible. extensions.conf [default] exten => 101,1,Background(tt-monkeys) exten => 1000,1,Dial(Sip/200,20,t,r) exten => 1000,2,Voicemail(s1000) exten => _XXX,1,Dial({Zap/g0/{EXTEN:1}) [telewest_pstn] exten => s,1,Dial(Sip/200,25,t,r) exten => s,2,Voicemail exten => s,3,Hangup zapata.conf [channels] context=telewest_pstn switchtype=national signalling=fxs_ks rxwink=300 channel => 2-4 adsi=yes usecallerid=yes cidsignalling=bell callerid=asr...

I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.....) [custom-inbound] exten => _556,1,answer exten => _556,n,playback(beep) exten => _557,1,answer exten => _557,n,playback(beep) exten => _558,1,answer exten => _558,n,playback(beep) exten =>

...> 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten => 265-NOANSWER,1,Busy [default] exten => s,1,Answer exten => s,2,Congestion The asterisk on the described side is connected to a classic company pbx (from Ericsson) and has local Extensions _XXX. The CAPI Interface gets signalled with MSN 260-265. Yesterday I rewrote the dialplan to allow calls from any company phone (_XXX) to 265 will be rerouted to PoC/11 (the IAX peer). This works fine, the IAX peer will be called with MSN 265 as callerid, so the called party can see the number for...

Hi. I've recently installed Asterisk on my Linux system and added the asterisk-h323 package. I'm working with a H323 plataform with gatekeepers and gateways. I'm trying to implement a Voicemail for the endpoints, that works when the endpoinst are BUSY. I'm a newbie in asterisk so i need a little help here... 1.- I have succesfully route the BUSY calls from my endpoints to my