How to use Attended Transfers?

[macafee]

Could you tell me how to use Attended Transfers?

[Philip Mullis]

hit the attended transfer key combo, you should hear transfer, mash in the extension/number of the person you want to call and then hit pound and it will ring/join up the call while your still on with the callee (unlike blind transfers, which are easy to remember as the fire and forget method of transferring)

[DrCain]

Can't get this to work.
When doing a blind transfer I first push ## , it says "transfer", after that I try to input the extension number. But it fails on the first digit telling me to try again and on the second digit it falls back to the call. When doing an attended transfer I don't even get that far, it just falls back to the call.

Any help would be appreciated.

May 22 22:14:39 asterisk[16836]: VERBOSE[17104]:     -- Starting simple switch on 'DAHDI/4-1'
May 22 22:14:49 asterisk[16836]: VERBOSE[17104]:     -- Executing [102@ANALOG-PHONE-20962721474be447077012e:1] NoOp("DAHDI/4-1", "internal calling internal phone: Phone 2 - <102>") in new stack
May 22 22:14:49 asterisk[16836]: VERBOSE[17104]:     -- Executing [102@ANALOG-PHONE-20962721474be447077012e:2] Set("DAHDI/4-1", "NOVOICEMAILWHENBUSY="yes"") in new stack
May 22 22:14:49 asterisk[16836]: VERBOSE[17104]:     -- Executing [102@ANALOG-PHONE-20962721474be447077012e:3] Gosub("DAHDI/4-1", "macro-main,s,1(SIP/102,102,102,tTo,)") in new stack
May 22 22:14:49 asterisk[16836]: VERBOSE[17104]:     -- Executing [s@macro-main:1] Dial("DAHDI/4-1", "SIP/102,,tTo") in new stack
May 22 22:14:49 asterisk[16836]: VERBOSE[17104]:   == Using SIP RTP CoS mark 5
May 22 22:14:49 asterisk[16836]: VERBOSE[17104]:   == Using SIP VRTP CoS mark 6
May 22 22:14:49 asterisk[16836]: VERBOSE[17104]:     -- Called 102
May 22 22:14:50 asterisk[16836]: VERBOSE[17104]:     -- SIP/102-00000004 is ringing
May 22 22:14:51 asterisk[16836]: VERBOSE[17104]:     -- SIP/102-00000004 answered DAHDI/4-1
May 22 22:15:04 asterisk[16836]: VERBOSE[17104]:     -- Started music on hold, class 'default', on SIP/102-00000004
May 22 22:15:04 asterisk[16836]: VERBOSE[17104]:     -- <DAHDI/4-1> Playing 'pbx-transfer.ulaw' (language 'en-us')
May 22 22:15:08 asterisk[16836]: VERBOSE[17104]:     -- Unable to find extension '1' in context 'macro-main'
May 22 22:15:08 asterisk[16836]: VERBOSE[17104]:     -- <DAHDI/4-1> Playing 'pbx-invalid.ulaw' (language 'en-us')
May 22 22:15:12 asterisk[16836]: VERBOSE[17104]:     -- Stopped music on hold on SIP/102-00000004
May 22 22:15:21 asterisk[16836]: VERBOSE[17104]:     -- Executing [h@macro-main:1] GotoIf("DAHDI/4-1", "0?vm-h,1") in new stack
May 22 22:15:21 asterisk[16836]: VERBOSE[17104]:     -- Executing [h@macro-main:2] Hangup("DAHDI/4-1", "") in new stack
May 22 22:15:21 asterisk[16836]: VERBOSE[17104]:   == Spawn extension (macro-main, s, 1) exited non-zero on 'DAHDI/4-1'
May 22 22:15:21 asterisk[16836]: VERBOSE[17104]:     -- Hungup 'DAHDI/4-1'

[niceguy_1920]

im not an expert or anything but i have been fighting with a similar problem. Have you taken a look under advance settings at the call limit for each extension? you may want to increase the default to 3.

[litnimax]

You should make sure that extensions are included. It simply does not find extension.

[DrCain]

im not an expert or anything but i have been fighting with a similar problem. Have you taken a look under advance settings at the call limit for each extension? you may want to increase the default to 3.

I must be blind or something, I really can't find it.
Running AskoziaPBX 2.0

litnimax, where do I check that?

[litnimax]

Seems there is a bug. Let me investigate. I'll come back soon.

[Alexcr]

If Num A call to B , and num B transfer to C all work OK
if num A set transfer , Askozia don't find context.

Unable to find extension '1' in context 'macro-main'
Context "macro-main" don't include extension or parking slot.....

In default Transfer is need for incoming call.

[litnimax]

Issue located. The problem is as follows.
Calls are handled in [internal] context, where flags tT are added. But actual 'Dial' command is called from [macro-main] context  using 'Gosub' application. That means TRANSFER_CONTEXT=macro-main, where only 's' extension is defined.

I think some time ago it was not 'Gosub' but 'Macro', and it worked well. Recently 'Gosub' app introduced a new option - parameters and started to behave like Macro. So I guess Michael replaced Macro with Gosub (because Gosub is more efficient than Macro), and thus broke transfers :-)
So, the cure.
Quick - place TRANSFER_CONTEXT in [globals] section in extensions.conf (press Esc, and open file editor).
Right - a) Change Gosub to Macro in extensions.inc b) Change TRANSFER_CONTEXT in dialplan before Dial.
Any other suggestions?

[DrCain]

Alright I am going to try this.
Also can we get this fixed in version 2.0.1?

[DrCain]

Any news on this?
Transfers seems to end up in [macro-main] even though all my phones are in [internal]

How do I fix this?

Michael Iedema really needs to look at this.

Maybe we should move this tread over to bug reports?

[Michael]

The transfer issue will be fixed for 2.0.2, to be released this month.

Thanks for the reports and research!

[netwalker70]

If Num A call to B , and num B transfer to C all work OK
if num A set transfer , Askozia don't find context.

Unable to find extension '1' in context 'macro-main'
Context "macro-main" don't include extension or parking slot.....

In default Transfer is need for incoming call.

Hi all, I'm new on this forum so I hope not to be OT...
Wainting for 2.02 release I'm testing beta r1732 with the same scenario:
now incoming calls can be correctly transferred, but seems if Num A calls an external Number, Num A cannot transfer to C.
Could be something wrong form my side?
Thanks alot for your help.