search for: dialoptions

Displaying 14 results from an estimated 14 matches for "dialoptions".

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2010 Jul 23
2
application call to Gosub affects flow of control, and needs to be re-written using AEL
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
2007 Oct 22
0
bristuff: music on hold but no dialoptions tT defined.
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]
2004 Nov 26
4
*67 or *57
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
2009 Jul 20
1
callforward with asterisk-gui.problem with stdexten
...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(${ARG1},b) exten = s-BUSY,2,Goto(default,s,1) exten = _s-.,1,Goto(s-NOANSWER,1) exten = a,1,Voicemai...
2009 Nov 25
0
Where are documented channel-dependant Dial options ?
Hi, I've recently discovered Dial examples such as "Dial(DAHDI/g4d/${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 --------------
2010 Nov 04
0
ring delay and DTMF related problem in asterisk 1.6.2.6
...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 som...
2009 Sep 18
0
Blind Transfer Won't Hangup
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
2015 Nov 04
4
Find me macro - calling multiple people to get a hold of one
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
2009 Jun 04
2
broken pipe in perl agi
...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)...
2005 Feb 24
0
Queue Questions
....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' logg...
2007 May 15
1
Asterisk is not showing the correct Incomming CallerID
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
2007 Dec 22
0
Dead Incoming call - Sangoma A200
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
2007 Oct 31
1
segfault - asterisk crash and restart
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
2007 Oct 23
0
Internal Data Stream Error
...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 mea...