Connection means asterisk reports transfer, if the user executes by the telephone set or the transfer command

Asterisk Asterisk Support
1

301/5000

Hello!!

I am new to the forum and ask for guidance.
I use asterisk-11 and 10 and I have two situation:

1 - change sip
2 - E1

What happens in the middle of the connection, asterisk reports transfer, if the user executes by the telephone device or the transfer command.

I am grateful for some guidance.

2

Both those versions are no longer supported.

More generally, I am having trouble understanding what you are trying to describe. Could you please provide log files (at least verbosity 5) and point out where in them things go wrong.

Please capture the log files from /var/log/asterisk, not from the console screen.

3

two marks:

first score with E1 signaling R2

I have the one context calling PSTN, where every external connection comes in.

Where I have extensions with SIP protocol.

No exten is like this:

exten => _X., 1, NoOp (-> RECEPTION <-)
same => n, Dial (SIP / 200,90, rt)
same => n, Hangup ()

In programming only who can transfer the connection and extension 200, but out of nowhere asterisk sends message transfer without the
Extension user 200 executes * 8 to transfer the call.

second, connection between extensions

Where I have extensions with SIP protocol.

No exten is like this:

exten => _ [1-5] XXX, 1, NoOp (-> INTERNO <-)
same => n, Dial (SIP / {EXTEN}, 90, rTt)
same => n, Hangup ()

Here the same thing happens from the PSTN context, out of nowhere asterisk sends message transfer without the
users performs * 8 to transfer the call.

Thank you for your attention.

4

SIP has a native way of initiating transfers, which doesn’t involve sending a digit string.

To further complicate this, some devices will do an a attended transfer, then hangup the original call, rather than explicitly doing an unattended transfer. Also, the native method of doing an unattended transfer before answer is very different from that for doing it after answer. That means even a native SIP transfer can be done in more than one way.

The t options enables all transfers

You can disable native transfers using this option, quoted from sip.conf.sample:

;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                ; Default is enabled. The Dial() options 't' and 'T' are not
                                ; related as to whether SIP transfers are allowed or not.

although doing so can confuse the user if the phone has a transfer button.

In the case of native attended transfers, including unattended transfers implemented using attended ones, the PBX sees the call to the transferee as a second line call from the transferor, not as a transfer. The call is already established before any indication is received that it is a transfer.

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