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Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw) exten => s,n(afterdial),Goto(s-${DIALSTATUS},1) Granted the particular above example could probably be better written to simply modify $ARG2 based on...

...is taken as a comment starting character, isn't it? Even in AEL: FRED="PJSIP/outgoing/sip:fred at sip.example.com;transport=tcp"; The ;transport=tcp doesn't seem to survive. I can add it onto the end of the variable in the Dial() command: Dial(${FRED};transport=tcp,${timeout},TtWw); and it has the desired effect, but of course specifying at the point where the Dial command is sets the transport policy much too broadly. Cheers, b. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 488 byte...

...oved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/943833473 -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup("SIP/...

On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote: > On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote: > > > > I can add it onto the end of the variable in the Dial() command: > > > > Dial(${FRED};transport=tcp,${timeout},TtWw); [ the part you trimmed ]: and it has the desired effect, but of course specifying at the point where the Dial command is sets the transport policy much too broadly. > You escape the ; character by placing \ in front of it. IE: > > DIal(${FRIED}\;transport=tcp,${timeout},TtWw); So pe...

...on files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten =>_9XXXXXXXX,1,Dial(mISDN/1/${EXTEN},45,tTwW) exten =>_9XXXXXXXX,2,Hangup() exten =>_9XXXXXXXX,102,Hangup() [default] exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Dial(SIP/101,30,Ttm) misdn.conf: [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log bridging=no stop_ton...

Hello everybody. I am trying to configure an Asterisk on Debian with the Billion ISDN card. I am using mISDN. But when I call on the CLI apears this: -- Executing Dial("SIP/101-081805b8", "mISDN/1/943833473|45|tTwW") in new stack -- Called 1/943833473 P[ 1] empty_chan_in_stack: cannot empty channel 255 P[ 1] --> we have already send Release_complete == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup("SIP/101-081805b8", "") in new stack == Spawn ex...

...i, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) -- Format for call is ulaw -- IAX2/215-33 is ringing -- IAX2/215-33 answered Zap/2-1 Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw to accomodate errors in vario...

...n logic [macro-stdexten] ; ${ARG1}=Extension ${ARG2}=Device(s) to ring exten => s,1,NoOp(stdexten ${EXTEN}) exten => s,n,Set(cname=${CALLERID(number)}@asterisk.deskoptional.com) exten => s,n,Set(CALLERID(number)=${cname}) exten => s,n,Macro(psa-predial) exten => s,n,Dial(${ARG2},20,tTwW) exten => s,n,Set(savestatus=${DIALSTATUS}) exten => s,n,Macro(psa-postdial) exten => s,n,Goto(s-${savestatus},1);ANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER exten => s-ANSWER,1,NoOp(call was answered!!!!!) exten => s-CHANUNAVAIL,1,VoiceMail(u${ARG1}) exten => s-CONGESTION,1,VoiceM...

...[,1] [1,] 0.9600812 > cov(x[20:30]) Erreur dans cov(x[20:30]) : fournir 'x' et 'y' ou bien 'x' en matrice And this behavior is braking function clustIndex (when used with unidimensional data), from the package cclust, file Rindexes.R, lines 137-147: ttww <- function(x, clsize, cluster) { n <- sum(clsize) k <- length(clsize) w<-0 tt <- cov(x)*n for (l in 1:k) w<- w+cov(x[cluster==l,])*clsize[l] zttw <- list(tt=tt, w=w) return(zttw) } Any way around it...

Anyone know why chanspy was not included in asterisk distribution as of October. ? I tried patching my current 1.0 but seems the patches are for an older version. I posted a bounty of $250 to get this to work with the newest stable. Needs be able to monitor bridged sip calls with or without a monitoring beep. Thanks John Bittner Simlab.net

Hello, here is my problem, After looking at the mail archives, I found a description of the error I get when I use this package. At first I even tought that they were showing how to solve it. But the thing is that by saying "the programmer forgot drop=FALSE" doesn't show me how I should get rid of the problem I have looked inside the package very quickly and I found three

...=no [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup [incoming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Dial(SIP/101,30,Ttm) [outgoing] exten =>_94XXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW) exten =>_94XXXXXXX,2,Hangup() exten =>_94XXXXXXX,102,Hangup() zapata.conf: [channels] signalling=fxs_ks usecallerid=yes callwaiting=no threewaycalling=no transfer=yes cancallforward=yes ; valores validos 256(32ms),512(64ms),1024(128ms) echocancel=yes echotraining=yes echocancelwhenbridged...

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

...round(auth-thankyou) exten => #61#,3,Hangup [macro-call-forwarding] exten => s,1,Set(temp=${DB(CFIM/${ARG1})}) exten => s,n,GotoIf(${temp}?cfim:nocfim) exten => s,n(cfim),Dial(SIP/${temp}@ruter) ; Unconditional forward exten => s,n(nocfim),NoOp exten => s,n,Dial(SIP/${ARG1},20,tTwW) exten => s,n,Set(temp=${DB(CFBS/${ARG1})}) exten => s,n,GotoIf(${temp}?cfbs:nocfbs) exten => s,n(cfbs),Dial(SIP/${temp}@ruter) ; Forward on busy or unavailable exten => s,n(nocfbs),Goto(s-${DIALSTATUS},1) ; NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER ... [incoming] ; ; Incoming cal...

...t;Zap/94-1", "tech_people|tT|||300") in new stack -- Started music on hold, class 'default', on Zap/94-1 -- outgoing agentcall, to agent '1234', on 'Local/4000@agents-b24a,1' -- Executing Dial("Local/4000@agents-b24a,2", "SIP/4000|20|tTwW") in new stack -- Called Agent/1234 -- Called 4000 -- SIP/4000-12c8 is ringing -- Agent/1234 is ringing -- SIP/4000-12c8 answered Local/4000@agents-b24a,2 -- Agent/1234 answered Zap/94-1 -- Stopped music on hold on Zap/94-1 == Spawn extension (agents, 4000, 1) exi...

...9)/video=0x0 > (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729) > > The call goes out the callwithus trunk: > [Aug 2 15:00:31] VERBOSE[1315] pbx.c: -- Executing > [s at macro-dialout-trunk:19] Dial("SIP/2001-00000047", > "SIP/CallWithUs/**********,300,tTwW") in new stack > > And then this, no INVITE goes out to callwithus at all: > [Aug 2 15:00:31] WARNING[1315] chan_sip.c: No audio format found to offer. > Cancelling call to ********** > [Aug 2 15:00:31] VERBOSE[1315] app_dial.c: -- Couldn't call > SIP/CallWithUs/***...

...wer guard expired Mar 31 13:10:51 NOTICE[14902] chan_unicall.c: Unicall/8 event Accepted Mar 31 13:10:51 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Channel gains Mar 31 13:10:51 VERBOSE[31370] logger.c: -- Executing Dial("UniCall/8-1", "IAX2/ast2:iaxpalancar at 192.168.1.20/1||TtWw") in new stack . . . .....everything continues normally Thanks in advice for your time. Ivan Reyes Tejera -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080331/1197f5ab/attachment.htm

...t;Zap/94-1", "tech_people|tT|||300") in new stack -- Started music on hold, class 'default', on Zap/94-1 -- outgoing agentcall, to agent '1234', on 'Local/4000@agents-b24a,1' -- Executing Dial("Local/4000@agents-b24a,2", "SIP/4000|20|tTwW") in new stack -- Called Agent/1234 -- Called 4000 -- SIP/4000-12c8 is ringing -- Agent/1234 is ringing -- SIP/4000-12c8 answered Local/4000@agents-b24a,2 -- Agent/1234 answered Zap/94-1 -- Stopped music on hold on Zap/94-1 == Spawn extension (agents, 4000, 1) exi...

...g=no [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup [incoming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Dial(SIP/101,30,Ttm) [outgoing] exten =>_9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW) exten =>_9XXXXXXXX,2,Hangup() exten =>_9XXXXXXXX,102,Hangup() Command line: modprobe zaptel modprobe wcfxo modprobe wctdm Then I start Asterisk (asterisk -vvvc), and when I call to the analog line number, the console shows that: *CLI> -- Starting simple switch on 'Z...

...t;Zap/94-1", "tech_people|tT|||300") in new stack -- Started music on hold, class 'default', on Zap/94-1 -- outgoing agentcall, to agent '1234', on 'Local/4000@agents-b24a,1' -- Executing Dial("Local/4000@agents-b24a,2", "SIP/4000|20|tTwW") in new stack -- Called Agent/1234 -- Called 4000 -- SIP/4000-12c8 is ringing -- Agent/1234 is ringing -- SIP/4000-12c8 answered Local/4000@agents-b24a,2 -- Agent/1234 answered Zap/94-1 -- Stopped music on hold on Zap/94-1 == Spawn extension (agents, 4000, 1) exi...