similar to: DIAL Event, who picks up?

Dear, I'm currently using a Dial command with multiple destinations and channels eg: Dial(SIP/100&SIP/101) I simply would like to know, in real time during the call (from dial plan or AGI), who has picked up the call. Can I find this information in a variable somewhere ? Thank you for your help Patrick

I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

hallo, i trying to dial with a python script via the manager interface, it works ok but i would like to send a soud file name as a variable to the dialplan, so that i can call a number and send it a different soundfile i choose in my pyton script. the problem is, that the * dials correct and sends a sound but only if its hardcodet, the variable my script sends will not bee seen in the

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

I'm in the process of setting up a 'phone tree', and are running into some problems. My goal is for users to dial a phone number, the asterisk system picks it up, plays the greeting, and users can type whatever they want into the system. What actually happens is users dial the phone number, asterisk picks up and additionally goes off-hook on another line, plays the greeting and

svn trunk 31497 For the life of me, I can't get this :) I want to be able to catch the situation where the calling party hangs up *before* the call is connected to the called party. My dialplan is thus: macro DialExternal(exten) { Dial(Zap/G3/${exten},120,g,M(connected)); goto DialResult|r${HANGUPCAUSE}|1; Hangup(); }; But the goto dialresult is not executed: Executing

I am working on this app to dial two external numbers. The second is dialed after the first hangs up. I have simplified things down to: exten => 3852,1,Dial(zap/g1/3964,10,g) exten => 3852,2,Wait(2) exten => 3852,3,Dial(zap/g1/7757,10,g) exten => 3852,4,Hangup Here is the debug: -- Accepting call from '0000000000' to '3852' on channel 0/23, span 1 --

Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/510@default-6b6c,1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755]

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

Tzafrir Cohen wrote: > > The agent picks up the phone but neither the agent nor the caller > > > here anything. >So please provide a more complte trace and a the relevant partt of your >dialplan. > Here is the relevant part of the dialplan: [local] exten => 98,1,Dial(SIP/sguenther,20,tr) exten => 98,2,VoiceMail(98|u) exten => 98,3,hangup exten =>

I have the following dialplan (segment) that isn't working as I expected it to: exten => s,n,Dial(Zap/1&SIP/202&SIP/203,18) exten => s,n,Dial(Zap/1&SIP/201&SIP/202&SIP/203,42) The plan was to have SIP/201 added to the group of ringing phones after 3 or so rings. What ends up happening, though, is the Zap/1 phone STOPs ringing when the dialplan falls through to

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't worth holding--One who holds them shows his self-weakness." Contact info: hank@hanksmith.net Email: Same as MSN. ----- Original Message ----- From: "hank"

Godd evening! What happens if someone calls and asterisk doesn't "Answer()" itself, but another analog phone does? Can I somehow catch this situation in my dialplan. I have an ISDN line, but with it I got a box with an adapter for good old analog phones. This doesn't seems to be directly connected to the ISDN line asterisk sees. But somehow, it must know, that the call

Hi I'd like to allow users to transfer a call to a specific park number. This way, the receptionist can tranfer a call park for ext 100 at park number 7100 etc... It seems like this should be fairly simple using the Park(ext) app but it doesn't work for me. No matter what I extension I use, the system just picks the next available park number. I've simplified my dialplan for