search for: dialopt

...out,Agent/${AgentNum}); } } context outboundQueue2 { .... } // outbound call logging for queuemetrics: macro qmoutqdial( clid, dialstring, queue, agent ) { start_dial_time = ${EPOCH}; QueueLog(${queue},${UNIQUEID},${agent},CALLOUTBOUND,-|${clid}); Set(dialopts=gWKU(queuelog_connect_event^${queue}^${UNIQUEID}^${agent}^${start_dial_time})); Gosub(pstnRouting,${clid},1); end_dial_time = ${EPOCH}; verb = COMPLETECALLER; &queuelog_hangup_event(${queue},${UNIQUEID},${agent},${start_dial_time}); return; } // central c...

Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983]

How to implement some of the function into asterisk like *67 "call number blocking" or *57 "call trace" ? I'm connecting to sipura SPA3K outside line by dialing 9+number. Is there a way to get outside dial tone in SPA3K PSTN-Line by dialing "9"? How to program the extension? -- #Joseph

...re---------------------------- exten = s,3,Set(temp=${DB(CFIM/${ARG1})}) exten = s,4,Dial(Local/${temp}@default/n) ; Unconditional forward exten = s,5,Set(DB(lastcaller/${ARG1})=${CALLERID(num)}) ; Note the last caller ------ends here------------------------ exten = s,6,Dial(${ARG2},${RINGTIME},${DIALOPTIONS}) exten = s,7,Goto(s-${DIALSTATUS},1) exten = s,8,Macro(stdexten-followme,${ARG1},${ARG2}) exten = s-NOANSWER,1,Voicemail(${ARG1},u) exten = s-NOANSWER,2,Goto(default,s,1) exten = s-BUSY,1,Voicemail(${ARG1},b) exten = s-BUSY,2,Goto(default,s,1) exten = _s-.,1,Goto(s-NOANSWER,1) exten = a,1,Voic...

Hi, I've recently discovered Dial examples such as "Dial(DAHDI/g4d/${EXTEN})" but I wonder where I can get an uptodate doc. Is there any CLI option such as "core show channel dialoption" that would explain what "g4d" exactly means ? "core show application dial" doesn't explain much about those channel-dependant Dial options. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/ast...

...ier, I could just not hear anything except the ring. that means other system picked UP a call and my SIP phone still here RINGS when i get connected it give me that my IVRS is started and some welcome prompt are also goes and once i connected i got prompt for entering something. is this due to dialoptions I passed '*rt*' or something version related issue with asterisk, also i note-down one thing that once my IVRS received call from my asterisk machine i am getting SIP 183 'session progress' not 200 OK for INVITE , Please help me to solve out this suggest some DIAL OPTIONS or...

...parties.agi: dbset CALLTRACE/38532 to 8688 > dialparties.agi: extnum 48532 has: cw: 1; hascfb: 0 [] hascfu: 0 [] -- dialparties.agi: dbset CALLTRACE/48532 to 8688 -- dialparties.agi: Filtered ARG3: 8532-38532-48532 > dialparties.agi: NODEST: 8532 adding M(auto-blkvm) to dialopts: trwTWM(auto-blkvm) > dialparties.agi: NODEST: 8532 blkvm enabled macro already in dialopts: trwTWM(auto-blkvm) -- <Local/28688 at from-internal-d4cc;1>AGI Script dialparties.agi completed, returning 0 -- Executing [s at macro-dial:7] Dial("Local/28688 at from-internal...

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

...after checking sippeer) exten => s,n,Set(LINESTAT=Idle) exten => s,n,AGI(hintcheck.agi|${ARG1}) exten => s,n,Wait(3) exten => s,n,Verbose(status is ${LINESTAT}) exten => s,n,Gotoif($["${LINESTAT}" != "Idle"]?inuse) exten => s,n,Dial(${ARG2},${RINGTIME},${DIALOPTIONS}) exten => s,n,Goto(s-${DIALSTATUS},1) exten => s,n,Macro(stdexten-followme,${ARG1},${ARG2}) exten => s,n,Background(vm-goodbye) exten => s,n,Hangup exten => s,n(inuse),Voicemail(${ARG1}) exten => s,n,Followme(${ARG1},${FOLLOWMEOPTIONS}) exten => s,n,Voicemail(${ARG1...

....agi: context = macro-dial -- dialparties.agi: type = Local -- dialparties.agi: rdnis = unknown -- dialparties.agi: enhanced = 0.0 -- dialparties.agi: dnid = unknown dialparties.agi: Caller ID name is 'Craig' number is '2203' dialparties.agi: Timer is 15 and dialoptions are tr Group is NOGROUP -- dialparties.agi: Added extension 2204 to extension map -- dialparties.agi: Extension 2204 cf is disabled -- dialparties.agi: Extension 2204 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin'...

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

...G[15840] db.c: Unable to find key '206' in family 'CFU' Dec 21 13:43:48 VERBOSE[15840] logger.c: > dialparties.agi: extnum 206 has: cw: 1; hascfb: 0 [] hascfu: 0 [] Dec 21 13:43:48 VERBOSE[15840] logger.c: > dialparties.agi: NODEST: 600 adding M(auto-blkvm) to dialopts: M(auto-blkvm) Dec 21 13:43:48 VERBOSE[15840] logger.c: > dialparties.agi: NODEST: 600 blkvm enabled macro already in dialopts: M(auto-blkvm) Dec 21 13:43:48 DEBUG[15847] manager.c: Manager received command 'Logoff' Dec 21 13:43:48 VERBOSE[15847] logger.c: == Manager 'admi...

...\000`\2074?7\000\000\000??u\000\000\000\000\000P?D\017\000\000\000\000?\234B\000\000\000\000\000?`zA\000\000\000\000??G\000\000\000\000" runningdata = "SIP/${fromsip_sippeers_hostid}${billsip_rt_routeprefix}${tosip}@${outbound_${billsip_rt_gw}}|${RINGTIMEOUT}|L(${availCallLimit}:120000)${DialOpt}\000zA\000\000\000\000\000\200a@\000\000\000\000P?zA\000\000\000\000?FM\000\000\000\000\000P?zA\000\000\000\000\020]zA", '\0' <repeats 12 times>, "?`"... oldargs = {0x0 <repeats 81 times>} argc = 1 x = 1098539312 res = 0 oldexten = "4612", '\0...

...18. Re: [France CID] Does Zaptel support ETSI FSK? (Jared Smith) 19. Re: Authenticate by IP? (Carlos Chavez) 20. Re: Extensions.conf for basic IVR? (Vincent) 21. Re: Authenticate by IP? (joakimsen at gmail.com) 22. Re: Authenticate by IP? (Victor Toofic) 23. bristuff: music on hold but no dialoptions tT defined. (Thomas Winter) 24. Re: [France CID] Does Zaptel support ETSI FSK? (Vincent) 25. Re: G729a codecs + Asterisk 1.4.11 (bilal ghayyad) 26. Re: [France CID] Does Zaptel support ETSI FSK? (Ira) 27. Voicemail playback on iPhone (Jason Lixfeld) 28. NAT traversal packet loss...