Asterisk and Re-Invite

Hello everyone!
I have a problem with Asterisk 10.9.0-rc1 and call hold
My problem is that Asterisk does not send the re-invite when I take a call from hold.
I’ve already entered canreinvite = no in all my SIP channels, set dtmfmode = info in sip.conf and my Dial () command you need to enter as an option t, T “,” h “,” H “,” w “,” W “or” L "(with multiple arguments).
I 'm follow this topic: … anreinvite
I debugged the asterisk, and I see that the re-invite are made ​​by asterisk, but in the SIP Header, the field to is composed by the IP address of the asterisk server, and not the next hop.
This is the log:
The asterisk’s machine has the following ip:
The next hop has this ip:
It is a bug? I have already searched on google, but I have not found anything
thanks for all
best regards

There are no packets sent by Asterisk in the pastebin!

SIP To headers are DNIS information, not routing information; that is based on the method line.

I don’t believe any official version of Asterisk generates outging re-invites when the hold status changed; that would be considered intended behaviour, if you tried to claim a bug.

ok,but it’s possibile to configure asterisk to send the re-invite that has received?

Best Regards

Asterisk never sends SIP packets it receives. It can regenerate them in some cases, and a change in destination, without a hold, when native bridging, would be a case where it would have to do that.

Asterisk is a back to back user agent, so it terminates the incoming dialogue and generates a separate outgoing one.

If it gets a hold indication incoming, it generates an internal AST_CONTROL_HOLD message, which causes MoH to be turned on on the outgoing side, but does not cause an outgoing a=sendonly to be sent.