similar to: Get Queue values from dialplan (Was: queue_variables() function)

Hi, Can somebody please shed some light on how to use the QUEUE_VARIABLES() function? The built-in help says ---cut--- Return Queue information in variables [Description] Makes the following queue variables available. QUEUEMAX maxmimum number of calls allowed QUEUESTRATEGY the strategy of the queue QUEUECALLS number of calls currently in the queue QUEUEHOLDTIME current average hold time

Hello, Has someone successfully used this QUEUE_VARIABLES() function (in 1.6.2-rc7) ? I tried to use it as I'm using SIPPEER() but without success. A previous question about it remainded unanswered ( http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the

All, I am trying to set a monitor file from the queue.conf as specified on http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf In order to avoid the default MONITOR_FILENAME format wich is: "agent-xxxxx-uniqueid.wav" for example "agent-10017-1266438575-26.wav" As you may now, when using the queue command you are not able to know which agent will take the call,

Hi, There are 3 new settings (setinterfacevar, setqueueentryvar, setqueuevar) and membermacro settings in 1.6 queues.conf. What is the potential use of these settings? The variables set are useful, but there is no indication of the purpose they could be used? Any one with some light on potential use case of these new features? raj

Hi I have this in my queue extension and I see this in asterisk when I call to the queue, but no file is created in the directory any ideas? exten => s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID}) -- Executing [s at kundservice:1] Set("SIP/0850001175-b7942770", "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12") Regards

Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if above screenshot from wireshark does not show here is a link for it: http://img402.imageshack.us/i/tempe.jpg/ I've tried a variety of setups in my IAX.conf (they all end up with the same issue, tried just bindaddr=0.0.0.0 with

2009/3/17 Olivier <oza-4h07 at myamail.com> > > > 2009/3/16 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I'm rather new to this domain so I may be doing stupid things without >> being concious of that. >> >> I've got a Patton MATA I'm trying to setup as T.38 fax adapter. >> Whenever I connect a fax machine (Dell

2009/1/27 Olivier <oza-4h07 at myamail.com> > > 2009/1/27 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I carefully followed instructions in README file lasting with : >> /root/register >> ... blabla >> asterisk -r >> CLI> restart now >> >> Then asterisk -r fails with : >> # asterisk -r >> Asterisk

Sorry for reposting, but even after extensive search I still did not find any answers. using: summary(aov(pointErrorAbs~noOfSegments*turnAngle+Error(subj/(noOfSegments+turnAngle)), data=anovaAllData )) with subj being a random factor and noOfSegments and turnAngle being fixed factors, I get the following results: ---------------------------------------------- Error: subj Df Sum

Hi Bill, > There's a problem with your check-in for r72619 is causing "weak > external" symbols to appear in C++ code when it shouldn't. Take this > code for example, > > #include <stdexcept> > > void dummysymbol() { > throw(std::runtime_error("string")); > } > > The c'tor for std::string is emitted as code from

2009/2/26 Olivier <oza-4h07 at myamail.com> > I must add I tried spandsp0.0.6xxx as a warning message advised me to do so > (using 0.0.4 would be ok for me but current trunk doesn't allow this > anymore, it seems). > > > 2009/2/26 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> With 0.0.6pre3: >> # ./build.sh >> CMake Warning (dev)

On Dec 4, 2009, at 12:52 AM, Duncan Sands wrote: > Hi Bill, > >> There's a problem with your check-in for r72619 is causing "weak >> external" symbols to appear in C++ code when it shouldn't. Take >> this code for example, >> #include <stdexcept> >> void dummysymbol() { >> throw(std::runtime_error("string"));

Leg/Transaction Does Not Exist" and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at

Hi, EveryOne: I am travelling CFG with MachineFunction. So I want to sure it. (Preccessors' Number) < MachineBasicBlock's Number < (Successors's Number), Is it really? best regards. ___________________________________________________________ 好玩贺卡等你发，邮箱贺卡全新上线！ http://card.mail.cn.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed...

Hi, I'm having alot of trouble understanding how to use dialplans for outgoing calls on Dahdi. Context : I have 3 TI spans, so 69 voice channels and three D channels (24,48,72). This is on a TE420B from Digium, if it matters. Here are my (apparently simple) questions in no particular order: 1) Dial(DAHDI/5555555555|20) doesn't work. But Dial(DAHDI/42/5555555555|20) does

2009/2/5 Olivier <oza-4h07 at myamail.com> > Hi, > > Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a > table listing ATA/Gateways combinations. > Could anyone successfully set a Patton M-ATA to work with another one, > using Asterisk 1.4 ? > > Is reinvite (canreinvite=yes) necessary or not ? > > Regards > > Replying to myself, I

Replying to myself, I've just read in 1.6.1 announcement that a new Incomplete dialplan application is the one that provides what I'm looking for ... 2008/10/3 Olivier <oza-4h07 at myamail.com> > Hi, > > If my memory serves me right, there was thread (in dev mailing list ?) > explaining how we could implement 2 stages dialing with SIP endpoints: > user dials 1234

2009/3/9 James Sneeringer <jsneerin at gmail.com> > On Mon, Mar 9, 2009 at 11:13 AM, Olivier <oza-4h07 at myamail.com> wrote: > > Anyway, whenever I'm typing make menuselect, app-fax is greyed out as in > my > > opinion, spandsp libriaries have not been found. > > > > Maybe, I should have typed something like (as suggested > >

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>