thanks for you all reply now i have a further knowing of SIP
to Samy which configure file should i add these syntax to ? root@CDlinux:/home/yliang1# ls /usr/local/opensips_proxy/etc/opensips/opensips_residential_2014-10-26_15\:6\:17.cfg */usr/local/opensips_proxy/etc/opensips/opensips_residential_2014-10-26_15:6:17.cfg* root@CDlinux:/home/yliang1# i followed the tutorial to install opensips to /usr/local/opensips_proxy On Mon, Nov 3, 2014 at 2:26 PM, Bogdan-Andrei Iancu <[email protected]> wrote: > Hi Michael, > > In SIP, each SIP server does server a certain set of SIP domains (defined > as FQDNs or IPs). Let's assume phone A registers with proxy A using domainA > and phone B registers with proxy B using domainB. > To have a call from phone A to phone B, A must dial B@domainB - so when > the call will land on proxy A, proxy A will see domainB is not locally > served and it will forward to proxy B (responsible for domainB). This is > call inter-domain DNS based routing and it is covered by the OpenSIPS > default script. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 31.10.2014 13:13, Michael Leung wrote: > > Hi all > > i know this is a stupid question > > but i dont use sip to make a phone call very often , > > i have setup up two opensips server in my intranet environment > > i use two phones to register on each server > > how to make a phone call from one to another one > > do i have to add the the destination domain name behind the alias number > when i dial out ? > > or why can i dial the alias number without domain name , then the > opensips server will routing it to a the opensips server automatically > > > thanks > > Michael > > > _______________________________________________ > Users mailing > [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >
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