Asterisk Attended Transfer

Asterisk Asterisk Support
1

i have problems getting the attendended transfer working on our Asterisk 1.2.12 system. Here is what i have setup.

In features.conf
[featuremap]
;blindxfer => #1 ; Blind transfer
;disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer

Using our SNOM 360 Phones, i have changed the transfer key to DTMF *2

In my extensions.conf I have

[macro-voicemail]
exten => s,1,Dial(${ARG1},${ARG3},rtT)
exten => s,2,VoiceMail(u${ARG2}@jtech-voicemail)
exten => s,102,VoiceMail(b${ARG2}@jtech-voicemail)

exten => 11,hint,SIP/11
exten => 11,1,Macro(voicemail,SIP/11,511,15)
exten => 12,hint,SIP/12
exten => 12,1,Macro(voicemail,SIP/12,512,15)
exten => 13,hint,SIP/13
exten => 13,1,Macro(voicemail,SIP/13,513,15)

Here is what happens :-

Extension 11 Calls Extension 12.
12 presses the transfer key and 11 is put on hold as expected.
When 12 Dials the Person to transfer to (13) the call is automatically hung up and 11 starts ringing through to 13.

Looks like a Blind Transfer to me.

I am trying for 12 to be connected to 13 announce the call, then when it is hung up by 12, 11 is talking to 13.

Can any1 shed some light on this for me??

Thanks,
Daniel.

2

Has anyone sucessfully setup Attended Transfers??

3

Hi Daniel,

Definately works for me on Asterisk 1.2.17 and Analogue handsets plugged into PAP2T’s.

Good luck…

4

Did you need any special settings to get it to work??

5

Hi Daniel,

Nothing special we (they - client) just dial the sequence as per the features.conf

I think they use *11, not sure if I set that or if thats what it was ‘out of the box’ - but it works!

seabro

6

I am slowly making progress with this issue. I should really slap myself in the face for not trying this sooner but anyway. Live and Learn.

While on a Call, if I press *2 on the SNOM Phone, I get the Attended Transfer and all works fine. However when i Program the SNOM Transfer button to send DTMF *2, it doesnt work. Looking at the SIP Debug, it looks like the SNOM is sending a SIP invite for another extension where as when I dial *2 it does not.

ANyone have an ideas on what may be causing this?? DTMF works on the phone because I can access external and Internal IVR’s and navigate them correctly.

Thanks,
Daniel.