similar to: After transfer context

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've created a tiny dialplan to test the return of a call on transfers, like this: (I had to use the DEVSTATE backport here) [phones] exten => _12XX,1,Dial(SIP/${EXTEN},6,tT) exten => _12XX,n,GotoIf($[ "x${BLINDTRANSFER}" = "x" ]?noBT) exten => _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)}); exten => _12XX,n,Goto(dRet) exten => _12XX,n(noBT),GotoIf($[

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, 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'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, 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.

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

On Wed, Jul 22, 2020 at 12:44 PM Administrator <admin at tootai.net> wrote: > No one on this ? > > Le 10/07/2020 à 18:06, Administrator a écrit : > > Hi, > > > > On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START > > and BRIDGE_END events > > > > zone-s*CLI> module reload cel > > The module 'cel' reported a

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?

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

I allow our internal extensions to transfer calls, so I have the appropriate "t/T" in the Dial() command. When I do the transfer, though, I don't know what context the user entry is interpreted in... The one which calls the macro which does the dial? This is an issue because my internal phones are in the 201-208 range. In my initial context, though, I only match on the

Hi all, Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf. It shows we can use variable BLINDTRANSFER to call back the one who transfer the call. However, in my tests below. The result is not as expected. case 1: A calls B (dial(sip/B||Tt) B answers and connects to A B transfer to C C doesn't answer the call and B ring again case 2: A calls B (dial(sip/B||Tt) B

Hi, How can you add specific statements into Asterisk dialplan (extension.ael, ...) for attented transfers ? I can see Asterisk sending Transfer or Masquerade events through AMI (in 1.6.1) but I could use an external program to catch those events but I would prefer to use dialplan instead. Any idea ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've found on wiki that there is a variable called ${BLINDTRANSFER} which should contain the channel (or a number) of user that made a blind transfer of a call to another extension. Also I've found a patch for chan_sip to add support for ${BLINDTRANSFER}, but it's not working at all (chan_sip crashing), so I guess it is intended for CVS-HEAD version. Has anyone tried to backport it

Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has

Recently I put callwaiting=yes in zapata.conf because customers want to speak to the operator in person, not leave her a voicemail, when she's busy with another caller. But now she can't transfer either of the calls (which she can do when there's only a single call). The operator has an analog phone connected to a TDM400B FXS line. The calls are coming from PSTN lines connected

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This might sound strange, but is there anyway for Asterisk to set extra sip headers based on a sip phone returning a 302 in a dialplan? Example: PSTN => Asterisk => SIP-Phone, SIP-Phone returns 302 Redirect, Asterisk sets X-Something: Some_Value & X-Somethingelse: Some_Other_Value, then sends the new invite with added headers. Stu Sheldon

Hi, on an Asterisk 16 with PJSIP I want to know the state of a device (idle, busy, unavailable, ...) in the dialplan. I tried with ChanIsAvail() but this one doesn't return the real state (eg a device calling an extension which is running ChanIsAvail() is marked as idle!) When I use in a console "core show hints" or "core show hint <EXTENSION>" I get the right

Hi list! When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 "Moved Temporarily"? The thing is that I'm trying to transfer incoming call from E1 interface back to E1 interface. Transfers will occur when user is going out and sets up all call

Greetings, folks. I'm having a problem with blind transfers. It seems that, despite not having the T flag set, callers are able to use the blind transfer option. Scenario is this: - Asterisk 1.2.14 - Caller calls into our call center on one of our many phone numbers. - Call gets placed into queue. - Operator answers call. - Caller is able to hit our blind xfer key sequence (#0) and dial