search for: dailplan

Hello I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangup() All is fine. Hello world is Playback and I hear a ring tone. If I remove the Playback hello-world. No ring

Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints "Unable to locate host" and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV -------------- next part -------------- An HTML attachment was...

Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digiu...

With softphone I mean linphone csipsimple or whatever. How should a dialplan lokks like? On 31.01.19 11:26, Antony Stone wrote: > On Thursday 31 January 2019 at 10:59:01, basti wrote: > >> Hello I use this dial paln: >> >> [o2-in] >> exten => o2,1,Answer >> exten => o2,n,Playback(hello-world) >> exten => o2,n,Ringing >> exten =>

This has been on my ToDo list far too long. I have a small call-center setup, with basic time of day/day of week validation before putting callers in the queues. With the holidays upon us, I need to add check to see if 'today' is a holiday so I do not put callers in unmanned queues. Due to how the agents work, I have to allow joinwhenempty. Does anyone have a snippet of dialplan code,

I want to add things to the prompts like: "mark urgent" "add to message" "pause while recording message" Any examples of how to do this? I'd also like to switch around prompts, not simply edit the sound files. Is it an agi, special dailplan, patching the app_voicemail.c file? All three? Any input/examples are welcome. -thanks __________________________________ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/

Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista -------------- next part -------------- An HTML attachment was scrubbed... URL: <htt...

Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by L...

...n number) off calls comming FROM a queue. This is my setup: - ISDN using zap (zaphfc) - Incomming calls arrive in a queue - One of the members of the queue is my cell phone... member => Zap/g1/XXX73XX19 (X to protect my privacy ;-)) The problem is the call from the queue to my cell phone. In a dailplan i can set the outgoing msn using SetCallerID, in queues.conf this doesn't work, resulting in a call without callerid (Dutch KPN using main number). Is there a way to set the outgoing msn (CallerID) of a call comming FROM a queue? Regards, Steven Lam

hi,all as you know, we can use MYSQL command to visit mysql database but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ? if not, is there any other alternative could do the same function(visit database in dailplan)? Thanks! -- Best regards, Sucan

I tried putting this as the ata's dailplan: *St4-|#St4-|9|^9t4>$.- this is sip.conf [ata2001] type=friend username=ata2001 secret=SoMeSeCrEt host=dynamic context=fromata canreinvite=no and this in extensions.conf [fromata] ignorepat => 9 exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten...

hi, All one thing confused me a long time. when i change the extensions.conf file. why not take effects after restart the asterisk? details as follow: my dailplan is : [95040] exten => _95040XXXXX,1,Set(CALLINNUM=${CALLERID(dnid)}) exten => _95040XXXXX,n(start),Answer exten => _95040XXXXX,n(welcome),Background(${welcomefile},,123) ... exten => i,1,Playback(invalid) exten => i,2,Goto(${CALLINNUM},welcome) first, i have changed _95040XXXXX...

...geCount(4000,count ).now app2: MyConfbridgeCount will call function count_exec(struct ast_channel *chan, const char *data).But at compile time char * data cause core dumped. Asterisk-11.5.1 Centos6 app_confbrige.c confbridge.conf ===================================================== Task: Using Dailplan user want to retrive no of user in conference '6050' => 1. Verbose(3,"testMyConfbridgeCount") [pbx_config] 2. MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,"== ${count} ====") [pbx_config] issue:Currently asterisk core dumped as soon as app2 load ....

...ng x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified callprogress=no is in place, and monitoring the two pstn lines with an external analog phone (with line lamps) indicates no one was on the phone and no ringing actually occurred. 2. Incoming CallerID seems to be slightly less reliable on the FXO compared to the x100p. (Eg, th...

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

Hi, I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service. I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. Two problems arise: First: How can I say to asterisk to get the message? (I see on CLI console that

Hi, I can't read a environment variable in a asterisk dialplan. When logged in as user root on the system an 'echo $HOSTNAME' gives the hostame of the machine. Asterisk (1.4) is started from the same console. I try to read it like this: exten => s,n,NoOp(host=${ENV(HOSTNAME)}) Does anyone know what i am missing? Ipv een saaie e-mail een leuk videobericht? Ga naar

...=========================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050' => 1. Verbose(3,"testMyConfbridgeCount") [pbx_config] 2. MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,"== ${count} ====") [pbx_config] ====================================================================...

...=========================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050' => 1. Verbose(3,"testMyConfbridgeCount") [pbx_config] 2. MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,"== ${count} ====") [pbx_config] ====================================================================...

Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near