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: voip-info.org/wiki/view/Aste … 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: pastebin.com/ARUC0j4t
The asterisk’s machine has the following ip: 184.108.40.206
The next hop has this ip: 220.127.116.11
It is a bug? I have already searched on google, but I have not found anything
thanks for all
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?
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.