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. 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:[email protected]/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.
incoming.xml.gz
Description: GNU Zip compressed data
outgoing.xml.gz
Description: GNU Zip compressed data
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