search for: dialoption

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call

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,Voicema...

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/asteri...

...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 so...

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

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},u...

....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' log...

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

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Hi all, Recently, I have upgraded the asterisk as following. asterisk-1.4.13 asterisk-addon-1.4.4 libpri-1.4.1 zaptel-1.4.5.1 Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh After upgrade, the server get segfault randomly and asterisk crash and restart itself. I got 2 core dumps of the segfault. Based on the core dump, we can't figure out the root cause to

...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 me...