thanks Samy
i will do the test in the following coming days.
Michael
On 11/04/2014 12:15 AM, SamyGo wrote:
Hi Michael,
You can follow what Bogdan mentioned and your call will get routed to
the other OpenSIPS as long as the FQDN is resolvable. If your users
don't want to add the other domain while dialling then you can add the
static routing code logic in the opensips.cfg file and restart opensips.
Thanks,
Sammy
On Mon, Nov 3, 2014 at 5:06 AM, Michael Leung <[email protected]
<mailto:[email protected]>> wrote:
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] <mailto:[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 Developer
http://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
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