Connect asterisk to an external IP PBX / Problem with transfers

Hi everyone! And thanks for reading me!
It is kind of complicated to ask the question without telling you how the configuration is.

I am connected to a Nec central, from asterisk, with trunks, each sip trunk is emulating a nec’s extension.
For example, I emulate being the extension 200 of the nec, with a asterisk’s sip trunk
When I call ext 200 (in the nec system), it rings asterisk. In this way, I have several extensions (let’s say ext 200 to 210 from the nec)

Until then everything works perfect. My problem starts when from Asterisk I want to transfer to another Nec extension. (Nec send a call to asterisk, and Asterisk needs to transfer to another nec ext)
If I want to transfer from asterisk 200 to nec 300, I am using another trunk / internal pair for the transfer (with a follow me 300 #)

So there are 2 trunks in use until the communication ends (one for incomming, other for out), and this is killing me, because I do not have so many SIP licenses from the Nec

Just in case I add that I have no way to connect through a SIP trunk with more than 1 channel

So there is the question. Do I have a way to send a TRANSFER command through the trunk, which allows the channel to be released, and not to use another secondary trunk for the oucomming call?
How do I forward the transfer through the trunk so the transfer take place outside of asterisk, and the main PBX take the course?

I hope beeing clear in the explanation
Thank you very much for reading this

Best Regards!

SIP doesn’t have trunks and extensions. Are you using FreePBX, in which case you are on the wrong forum.

If using bare Asterisk, please provide your pjsip.conf (or at a push sip.conf) configuration that you consider constitutes a trunk.

There is no “follow me” abstraction as an Asterisk primitive. You are definitely on the wrong forum. Peer support for FreePBX is provided at

The AsteriskTransfer() application is used for initiating blind transfers, in both answered and unanswered state. Asterisk doesn’t support initiation attended transfers, or de-tromboning of signalling.

Sorry David, you were right, I’m working with freepbx
I started a new topic in this forum

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