[OpenSIPS-Users] how to make a phone call to between two different domain sip server

[Michael Leung]

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 at 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 at 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 <bogdan at opensips.org>
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 at 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 listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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