similar to: Parking problem with outgoing calls

Hello, i'm using the parking feature. When the call is parked by A (number *15) , B is correctly parked, by A did not hangup automatically. Here are the dialplan [local] exten => _XXX.,1,Wait(0) exten => _XXX.,n,Dial(SIP/${EXTEN:0}@trunk_sip_2,0,TK) exten => _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK) exten => _XXX.,n,Playback(callbox-thinkro-trunkDefautIndispo) exten =>

Hi everybody, I have a sip phone (Siemens) which has no sip functions at all. Is is possible to press #4 by example to put the call on hold then dial #2 to get the call back ? I'have look at features.conf but i did not find the solution. I know the call parking functionnality, but i would like a much simple system. I hope i'm clear enough. Thank you Matthieu NICAISE Responsable

Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the "please leave a message after the beep". I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system.

Hi, I have 3 ISDN lines using the digium B410P card. Incoming and outgoing call are working. I use the following version : * libpri-1.4.10.2 * dahdi-2.2.1.1 * asterisk-1.6.2.6 On incoming calls, DTMF is not working, i can't see any logs. Here are the main configuration files : dahdi-channels.conf ;; Span 2: B4/0/1 "B4XXP (PCI) Card 0 Span 1" signalling=bri_cpe

Hi, I've defined a feature using a macro in features.conf : special => #2,peer,Macro,special Everything is working if the user use the phone key. But i would like to call the feature (or the Macro on the peer channel) from AMI or CLI. First i thought i would be simple, but i did not find any solution. Does someone has an idea ? Thank you very much. Matthieu NICAISE Responsable

Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks???

I am trying to make a system where a conference user can invite others to join. I am running the 1.2 version of asterisk, so can't use the example on voip-info.org. With use of the X flag on a meetme conference to exit with a single digit, I can get people to join me in a conference with exten => _XXX,1,Dial(${THEIR_EXTEN},,dG(conference-context^${CALLERID}^1)) where the

Hi everyone, For some reason MixMonitor doesn't record when it should; It actually shows the MixMonitor line just fine on the CLI. How can MixMonitor be debugged for things like privilege issues or filename issues? **I had this working at one point and then stopped working. Not sure what I changed. System Info: Asterisk 1.4.21.2 Queuemetrics 1.6.3.0 [queuedial] ; this piece of dialplan is

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

I want to be able to pass any number (variable length) to a context and then forward that to another asterisk server for processing by that servers dial plan.? I have the two talking IAX2 so that part is done. I can also dial a number from the sending to the server asterisk. The problem is I don't want to have to create (duplicate) dial plans at originating Asterisk to equal those at the

Apologies if this is a resend, but I've sent this 12 hours ago and still can't see it on the list. Hi, I've just started to setup my phones with Buddy watch. Basically, it all works fine when using the simple example on the wiki: exten => 123,hint, SIP/some_sip_reg exten => 123,1,SIP/some_sip_reg BUT, what I need to do is dynamically decide where the hint checks for buddy

I have been trying to get the dynamic parking working. For some reason when I park a call using this method the console says it is using the default parking context not the one I am trying to specidfy. It also is playing the parked extension to the caller. I am transfering the call to an extension that is doing a goto to the context below. Any ideas or examples on how to make this work.

Hello list, I want to get the phone number out of the following P-Asserted-Identity header : /"BlaBlaBla" <sip://88779922//@192.168.8.10;user=phone>"/ I do the following in the dialplan : /exten => _XXX.,n,Set(PY=${SIP_HEADER(P-Asserted-Identity)}) exten => _XXX.,n,Set(PY2=${CUT(PY,@,1)})/ This gives me : /"BlaBlaBla" <sip://88779922/ How can I

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Now that I have a little more time, I was debugging my dialplan and it was of the following way: - ------------------------------------------------------------------------- ; DGB - 20090615 [macro-dial] exten => s,1,Dial(${ARG1},15) exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u)

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

I am trying to get two * boxes to communicate with eachother. I have read http://www.voip-info.org/wiki-Asterisk+-+dual+servers as well as information on IAX channels, the Dial() command, and the switch statement in extensions.conf. But I am having no luck. I have a working * box running with a Zap card that I want to be the server machine. I have another little box running * with just a