Re: [sipX-dev] sipxbridge: "Use public address for call setup" parameter works for outgoing calls only
On Fri, Jun 26, 2009 at 8:08 AM, Nikolay Kondratyev<kond@xxxxxxxx> wrote:
> Hi all,
>
>
>
> In the gateway configuration there is a “Use public address for call setup”
> checkbox.
>
> Here is the description: “If checked (default), use the sipXbridge public
> address for SIP signaling and SipXRelay public address for SDP (media
> description). Otherwise, the local (LAN) address will be used. Using the
> local address assumes that the ITSP provides NAT traversal compensation.”
>
>
>
> So if I use a “routable” (no NAT on the way to it) gateway, but want to use
> sipxbridge, I create a “sip trunk” gateway, set “route” parameter trough
> sipxbridge, and uncheck “Use public address for call setup” checkbox.
>
>
>
> I found that in this configuration sipxbridge really uses internal address,
> but only for outgoing calls. For incoming calls sipxbridge uses external
> address anyway. See attached files.
SipXbridge needs to know which ITSP account to associate the inbound
INVITE. Using this information, it can rewrite the addresses in the
OK response. To make this determination, sipxbridge uses the host and
port of the topmost via header to try to find the account. In your
configuration, you have a single "ITSP account" set up to the domain
yate4.lab.nstel.ru. Try specifying an IP address in the ITSP proxy
address ( you currently have yate4.lab.nstel.ru which is a domain name
). This should enable sipxbridge to find that account. Let me know if
that works for you.
Ranga.
>
> Here is my setup (I use Yate on the same machine as sipX as sip-h323 gateway
> to avaya IPOffice):
>
> SipX(172.23.12.104) ------sip------- Yate(172.23.12.104:5059)
> ------------h323-------IPOffice(172.23.14.2)
>
>
>
> In the incoming.xml file one can see (frame 22) that “Contact” and SDP part
> are constructed using external address – 81.211.30.104.
>
> I can hear bidirectional voice in both outgoing and incoming call, but for
> incoming call I actually don’t understand why voice is ok… Looks like it’s
> “internal Yate trick” …
>
>
>
> The whole snapshot is available at
> ftp://sipx:sipx@xxxxxxxxxxxx/sipx-configuration-sipx4.lab.nstel.ru.tar.gz
>
>
>
> I think it is a problem on the sipX side. Am I right? Should I create an
> issue regarding it?
>
>
>
> Thanks and regards,
>
> Nikolay.
>
>
>
> P.S. forgot to say : I use 4.0.1-015823 version.
>
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--
M. Ranganathan
