similar to: Asterisk action when transfer occurs

Hi, I will confess immediately that this is only tested on 1.2.24, and I would be interested to know if it happens on 1.4, but I cannot find a bug-tracker entry which represents this issue. Consider a PSTN call which comes into asterisk, and is bridged to a SIP phone. The phone operator then places the call on hold (hold music plays) and a second call is made from this handset to a Queue...

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

On 12/20/2014 03:22 PM, sean darcy wrote: > On 12/19/2014 09:42 AM, Rusty Newton wrote: >> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >>> I've got a confbridge set up which works if dialed locally: >>> >>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >>> --

On 12/21/2014 04:42 AM, Patrick Beaumont wrote: > Have you enabled DTMF logging and seen the DTMF codes being recognised by > Asterisk? I had a bunch of soft phones that I had to change to using ?sip > info? for the DTMF signalling as the RFC signalling was not always being > recognised. This would cause transfers to appear as if the user had not > dialled any digits. > > >

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised by >> Asterisk? I had a bunch of soft phones that I had to change to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear

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

Am 05.03.2015 um 01:09 schrieb James B. Byrne: > I am trying to determine how the transfer button on the Snom-870 works > with Asterisk. Is the ## special code employed as when it is entered > through the handset or is the blind transfer through the phone > function accomplished in a different fashion? > Hi, I hope I understood your question correctly. AFAIK, the transfer

Am 05.03.2015 um 15:09 schrieb James B. Byrne: > > On Thu, March 5, 2015 05:30, Ruben R?gels wrote: >> >> >> Am 05.03.2015 um 01:09 schrieb James B. Byrne: >>> I am trying to determine how the transfer button on the Snom-870 >>> works >>> with Asterisk. Is the ## special code employed as when it is >>> entered >>> through the

I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? -- *** E-Mail is NOT a SECURE channel *** James B. Byrne mailto:ByrneJB at Harte-Lyne.ca Harte & Lyne Limited

Hi, I am using a number of snom190 phones, and an asterisk "gateway" server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended transfer works fine if the originating phone call is NON-SIP

Hi BK - >> The blind transfer does not work. >> >> The way we try to blind transfer a call: >> 1. answer the call >> 2. press transfer >> 3. press blind softkey -> the display shows "Blind transfer to:" and >> cursor is in the second line >> 4. enter the number -> when we enter the second digit of the number >> the

On Thu, March 5, 2015 05:30, Ruben R?gels wrote: > > > Am 05.03.2015 um 01:09 schrieb James B. Byrne: >> I am trying to determine how the transfer button on the Snom-870 >> works >> with Asterisk. Is the ## special code employed as when it is >> entered >> through the handset or is the blind transfer through the phone >> function accomplished in a

I'm coming to Asterisk from a traditional PSTN environment, so forgive me if I use the wrong Asterisk/SIP terminology. I need to make a product where calls come in go through various menus and based on various configurations perform attended transfers, blind transfers, and patch callers together. For patching two calls together, my thought is that this would be a conference in

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

Hi all, I'm trying attended transfer on Asterisk 1.0.7 and AT-320 phone. I met a lot of problems during this steps, while in the blind transfer all works fine. I had this kind of problem: CASE 1: A call B B set on hold A B call C (that is busy for some reason) B try to get the first call with "hook flash" (or pressing the "hold" key) and A stop to work. B

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

Hi, I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer. My extensions.conf: [globals] __TRANSFER_CONTEXT = transfer [common] exten => 123,1,Playback(demo-congrats) exten => 123,n,Hangup() exten => _0X.,1,Dial(SIP/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten

I cannot seem to get any transfers to work at all. The console show I have #1 amd #2 set up for Blind and Attended Transfer, but when I hit these buttons on my handset nothing happens (other than I hear the dtmf tones on the other end of the line). roo*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8

Hi all, Is there a way to tell which extension transferred a call in a blind transfer? Sorry if it's a basic question, but I haven't seen an answer. ${CALLERID(num)} still holds the outside party caller id (which it should), but I'd like to the extension number of the extension that transferred the call. Any suggestions? Thank you, -- Warm Regards, Lee