similar to: Blind Transfer - Who transferred the call?

Hi, I want to check the status of a blind transfer (only sip endpoint) between various phones. Transfer is working perfectly, using ## from features.conf or using transfer key from phone, here SNOM320. My problem is that if party to transfer to is busy, the transfer fail and the call is ended. What I want to do is to return the call to the party who originate the transfer. I checked

Hi, I am having an issue under a specific circumstance with Asterisk 1.4.26.1, using blind transfer. If my SIP phone dials a number (so I am the caller, happens on both Polycom phones and eyeBeam softphone), do a BLIND transfer to another nuber (internal or external) ${CDR(accountcode)} is NULL fo the rest of the dialplan. My dialplan logic depends heavily on knowing the accountcode.

Hello Users, We have Thirdlane Multi tenant PBX system in production. Asterisk version is 1.6.2.15. Attendant transfer is working, but blind transfer is not working with Grandstream (gxp-2000) phone. We have read from google that it is a bug in Asterisk 1.6.2.15. We saw the below links: <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.o

Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks

Hi, I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? If there is an A->B call going on, I'd like to know which side did the transfer - but whichever side does it, I get back to context

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hi, Is there a way to do a blind transfer within an asterisk dialplan (like '##')? The reason I need this (I think) rather than a regular Goto() is that I'm trying to do one-touch parking. I can park a call using one-touch parking and then pick it up again, however if I try to re-park the call, it gets lost. I think that is because asterisk thinks I'm still on the park extension.

I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting "Explicit MWI Subscription" to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee

Hello Everyone, I can see that a few people are interested in SIP WIFI phones. I have tested several Linksys 300,and it is OK. More of a toy then a business tool. It a poor built in ear speaker, which makes all calls sound tinny, and the unit is known to hang. I have two Linksys 300's that are fun to play with however, I wont hand them out to users. HOWEVER- The Zultys WIP 2 is an

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

I'd like to transfer all my pesky telemarketing calls to Jolly Roger . http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html In the middle of a call I'd hit some DTMF sequence, which would dial Jolly Roger and transfer the call after Jolly Roger answers. But blindtransfer requires an extension after you hear "transfer". And I don't

Hi all, Running 1.2.14 When I call a FastAGI script such as this script for an incoming call: [calldirect] exten=>s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17

Hi Everyone, I'm wondering if anyone here write AGI's in compiled binaries. I'm writing a small Cepstral AGI in Freepascal/Lazarus. I know there are some other AGI's out there, but I wanted to add some more functionality than what is available such as having the AGI determine if the "data" argument is plain text or a path to a text file and act accordingly. The

While using Blindtransfer #Extension everything works fine. But how do i activate announced transfer with an Grandstream GPX2000 ? Greets Markus

Announcement: We are pleased to announce that we have released AsterPas FastAGI ObjectPascal Script Server for Asterisk PBX under xxxx license. What is AsterPas? AsterPas is a FastAGI server which allows real-time scripting of Asterisk PBX call flow using ObjectPascal based scripting. AsterPas includes many built objects available from scripts such as Cepstral TTS Engine class, database access

I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and

Dear all, I am trying to write an R function that can estimate Transfer functions *with additive noise* i.e. Y_t = \delta^-1(B)\omega(B)X_{t-b} + N_t where B is the backward shift operator, b is the delay and N_t is a noisy component that can be modelled as an ARMA process. The parameters to both the impulse response function and the ARMA noisy component need to be estimated simultaneously. I

Hi, When a SIP hardphone is transfering a call while ringing (caller and callee don't speak to each other) using phone's Transfer key, it seems BLINDTRANSFER remains empty. Though I can see a 302 MOVED TEMPORARILY message coming in. Is there a work around or something obvious I'm missing (it's the first time I'm playing with Dialplan transfert features. context mylocal {

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?