Hye everybody, anyone has any idea how to help me?
To resume, I just want to know how to change the IP in the URI sent by
Asterisk (first line of SIP packets)
Thanks for your time!
++
On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent lf...@leurent.eu wrote:
Hello All,
I have a little
Modify the $ru pseudovariable or use rewritehostport() out of core.
This is not the right mailing list. This belongs on the
OpenSIPS/OpenSER lists.
There is also a mailing list we operate called SER-Asterisk-Interwork
that is specifically intended to address SER* / Asterisk integration
Hello,
it is not an OpenSIPs problem I have, it's an Asterisk one,
I would like to change the URI in message generated by Asterisk.
Thanks
Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit :
Modify the $ru pseudovariable or use rewritehostport() out of core.
This is not the
The Request URI generated in an INVITE originated by Asterisk is
governed entirely by the parameters passed to Dial().
For example:
Dial(SIP/1...@peer_name)
... will generate a Request URI of
1...@host.or.ip.of.sip.conf.peer.named.peer_name.
It is also possible to send requests to hosts
Thank you, this is exactly what I needed!!
In order to Dial any number to a registered peer, I just have to enter
Dial(SIP/anynum...@sippeername)
Best Regards!
Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit :
The Request URI generated in an INVITE originated by Asterisk is
I have spoken to quickly,
Usually Asterisk on an incoming call sends an INVITE Reg.Contact
Number@Reg Contact IP to the Peer IP.
With the command you gave me, it is possible to send an INVITE
othernumber@Peer IP to the Peer IP.
What I would like to do is to send
Hello All,
I have a little complicated question about the Dial command.
I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on
Asterisk servers.
Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server.
Everything works except for trunk numbers:
For each
