similar to: announced transfer

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?

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

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

Hi, I'm playing around with the queue features, and I'm looking for a solution to indicate the agent status (logged on / logged off) on the phone (Grandstream GPX2000 or SNOM 190) It would be nice to use one of the busy lamps... thanks -- Alexander Topolanek Intelligente EDV- und Telekommunikationsl?sungen Montage von Sicherheitstechnik Probst Peitlstr. 85 2103 Langenzersdorf +43

I have a challenge that is ending up quite interesting. I need to identify which SIP phone "touched a call last", that is, which phone did the last transfer or dialed the original call if no transfers were done. It is easy in the case of a regular, non-transfered call. Just put something in callerid= in sip.conf, and that will show up in ${CALLERID}. The same with an attended transfer,

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

Well, I wake up this morning, and aussievoip isn't up. I ring godaddy, who _were_ hosting it, and they say that the machine's been compromised, and you can't have your data. Nyah Nyah. I spent 1 hour and 38 minutes on the phone to them, trying to convince them to let me somehow get access to it, but to no avail. I've reported it to the Australian Federal Police High-Tech Crime

Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus

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

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

OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an *@Home box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something

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

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 list, I'm wondering if it's possible to transfer a call that is still ringing??? I Have some Grandstream GXP-2000 and with the TRNF button it's impossible. So, I've configured some keys to transfer the calls like this: [featuremap] blindxfer => #2 ; Blind transfer (default is #) disconnect => *0 ; Disconnect (default is *) ;automon => *1

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 everybody, I'm not a programmer, so I really don't know this, but is it possible to somehow backport ${BLINDTRANSFER} variable functionality to 1.0.X versions of *. I need this really badly ... thanks for your replies. Ivan

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.

Hi I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility. I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to look ? I can attach conf files etc. if needed. Asterisk says it has 30 capi channels available,

Hi ! :) Does someone have an idea of the reason why my MusicOnHold (with clean mp3 downloaded from http://aussievoip.com.au/wiki-MOH) is always having interruptions and micro-cuts ? No high load of the system, no swap, no hard disk r/w, mpg123 seems running well... nothing ! Except a little message at startup : "Warning, flexibel rate not heavily tested!" I'm getting