search for: diaplan

I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Please help _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969...

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML attachment was scrubbed... URL: http:...

Hello, Is there a way to remove a global variable from dialplan ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090615/03d38624/attachment.htm

Hello, For an (old) Asterisk 1.4, how can I tell from the dialplan that a parking lot has available slots, or that a parking slot is empty ? Shall I just park the call with Park() application and , for instance, program next priority as if it would be triggered when the parking lot is full ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I think this have been talked over several times but I couldn't find any answer. Sorry for asking. I want from dialplan, to transfer a callee to a context-extension-priority that would play a given fax file to callee (callee is supposed to be a fax number). I can get caller's channel id (with built-in CHANNEL variable). I found BRIDGEPEER but its value remains unset (see bellow)

Hi, I have a little situation with my dialplan, and I am wondering if what I want is even possible. Here it is: I have three contexts, context1 includes contexts2, and context2 includes context3. In other words, in context1 all extensions of context2 and context3 are valid (and actually working, so that's good). I am using those context for the sake of code clarity and reuse, and for

We intend to use Asterisk with a very large dialplan (with a lot of functionality for 3000+ users). Each user will be able to change several of his parameters in the dialplan, so we will be forced to reload the diaplan constantly. Has anybody else any previous experience with a similar installation? There are some things that we'd like to know, if anybody can help us. These are: - Is this something that can be done safely with Asterisk? - Can we have a diaplan configuration update every 5 or 10 minutes...

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/piperm...

Is there an easy way to setup diaplan so when someone pushes a digit such as * during a call, they will be transferred to another destination. For example, a caller is hearing ringing while calling a UA, but instead of waiting for the UA to pick up, they can push * and go directly to that UA's voicemail. -------------- next part --...

...ible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an AGI/diaplan script that ask to accept the call pressing a specific key (OK) 4) if right key is pressed redirect the current call to the channel2Link, connecting the call in queue with the remote number (?) Step 1,2,3 works properly but i'm not able to link the two channels, even using redirect,goto or pic...

...o Asterisk 13 and can't figure this one out. I'm pretty sure the question has been already asked, but I failed to find a solution. Can you modify CDR values in an h-extension? My cdr.conf contains: [general] enable=yes unanswered=yes endbeforehexten=yes initiatedseconds=no batch=no The diaplan contains a simple "h" extension exten => h,1,NoOp(${CDR(userfield)}) exten => h,n,Set(CDR(userfield)=changed) exten => h,n,NoOp(${CDR(userfield)}) In the same context I execute: exten => 10,1,Set(CDR(userfield)=empty) exten => 10,n,Dial(SIP/10) The "h" extension...

Dear asterisk-users, I have listened that a diaplan on Asterisk can extract information from proprietary SIP messages header fields. That is, if Asterisk receives a SIP message with a modified HEADER (containing proprietary fields) , is it possible to program the dial plan to make Asterisk extract the values of such fields, being possible to handle...

...nge in the dial app of the new chan_capi; unfortunately I did not find (may be it's too late now for the two neurons still awake) the message or readme where the change was mentioned, even using heavy 'find scans' in my message database. No luck on the wiki, either... The old (reused) diaplan tries to dial out with: >exten => _3.,1,Dial,CAPI/<mynumber>:b${EXTEN}|90 > ; Cellular Phone numbers >exten => _3.,2,Goto(s-${DIALSTATUS},1) >exten => _3.NOANSWER,1,Hangup() ; Hangup >exten => _3.BUSY,1,Hangup() ; Hangup The console says: >Aug 3 23...

Does IAXy2 have any internal diaplan? What I need to do is have a phone that when picked up automatically dials out in a hotline fashion. I know that I can setup Sipura devices to do this, but I'd rather use IAX2. Alternatively, do any of the cheapish SIP phone support this sort of internal dialplan functionality? Thanks, Michae...

Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM about 75 Polycom Phones, one E1 for incoming calls. We have program a page system with the page command and the auto answer funtion of polycom. We have detect via diaplan if the phone isn't in call we place the call. All this via Macro. But in the our that they are not many calls. So much user that can be page.. The Asterisk crash. We think it is a RAM Memory problem.. Do you have any idea for this ? Thanks. -------------- next part -------------- An HTML at...

...6824]: misdn_config.c:579 _build_general_config: misdn.conf: "presentation=allowed" (section: general) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config: misdn.conf: "diaplan=0" (section: general) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". I don't understand because the parameters : - use_callingpres - presentation - dialplan are set in the correct section.. My misdn.conf : [general] debug=4 append_digits2ex...

Hi, In sip.conf, it's possible to add a line such as setvar=MYFIELD=foo and access this value from diaplan with SIPPEER function. 1. Which function is available to access values in users.conf such as vmsecret ? 2. Is it possible to extend users.conf with custom keys/values ? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/aster...

Hi, I am registering my Asterisk boxes to a SIP provider for outgoing calls. My "outgoing" dialplan context tries to dial out in sequence, starting with the SIP provider then ISDN lines and finally analog lines. So the idea is that if the SIP trunk fails then all calls are dialed out via ISDN and analog. I noticed however that if I switch my DSL connection off (ie. no internet access

Hello, When reloading a diaplan (asterisk 1.6.1.X), I can see in console : [Apr 2 09:02:00] WARNING[2217]: ael/pval.c:2522 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 621-621: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! W...

Hello, My setup is : phone ----- PSTN/ISDN ----- Patton SN4638 ------- Asterisk (Asterisk is in 1.6.1.18, Patton in 5.3) When I call the Asterisk, I can read from console that : - the call comes in, - the line MusicOnHold(,10) in my diaplan is reached and played, - I see RTP packets coming in and out (hundreds of lines such as: Got RTP packet from 192.168.102.200:4890 (type 00, seq 005360, ts 2343932047, len 000160) Sent RTP packet to 192.168.102.200:4890 (type 00, seq 036824, ts 082080, len 000160) ) - but I can't hear a...