ARI: PJSIP_TRANSFER_HANDLING=ari-only not preventing attended transfer

Asterisk Asterisk APIs
1

Hi,
I am using Asterisk ARI and want to handle warm transfers (attended transfers) manually. This means Asterisk should not complete the transfer by itself.

I found a solution for this by setting PJSIP_TRANSFER_HANDLING to ari-only in the dialplan, like this:

[from-twilio]
exten => _+X.,1,Verbose(Call from ${CALLERID(num)} to ${EXTEN})
same => n,Set(PJSIP_TRANSFER_HANDLING=ari-only)
same => n,Stasis(stasis-bifrost-dev,incoming)
same => n,Hangup()

When checking with core show channel <channel_id>, it shows:

sipmedia1*CLI> core show channel PJSIP/sipcore-00000060
...
PJSIP_TRANSFER_HANDLING=ari-only
...

So it seems PJSIP_TRANSFER_HANDLING is being set correctly.

However, when transferring the call, the transfer still completes automatically. I receive both a REFER and a 202 Accepted during the transfer.

I also tried setting PJSIP_TRANSFER_HANDLING to ari-only in the variables when dialing the second leg, but the same issue occurs.

Could you please help me fix this issue?

2

That’s not how you set it. The full documentation is here including how to set it and use the functionality:

It ultimately will still do a 202, though.

3

Thanks for your response. I will check the documentation.

4

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