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

[Michael Leung]

HI all

I Define the following regular expression in my first opensips1 domain
config file

    if ($tU =~"^3*") {

        #strip(1);

        if(!lookup("location")) {
            if(!t_relay("udp:opensips2:5060")){

            sl_send_reply("404", "User Offline/Not Found");
            }
        }
   }


and then Define another regular expression in my second opensips2 domain
config file


        if ($tU=~"^3*" && $fd != "opensips2") {

                if (!lookup("location")) {
                        sl_send_reply("404", "User Offline/Not Found");
                } else {
                        t_relay();
                }
        }

when i dial 30001 from 10001 at opensip1

i got error from opensips1


Nov  6 20:35:53 CDlinux /usr/local/opensips_proxy/sbin/opensips[25544]:
ERROR:dialog:get_routing_info: failed to print route records
Nov  6 20:35:53 CDlinux /usr/local/opensips_proxy/sbin/opensips[25545]:
ERROR:core:print_rr_body: too many RR
Nov  6 20:35:53 CDlinux /usr/local/opensips_proxy/sbin/opensips[25545]:
ERROR:dialog:get_routing_info: failed to print route records
Nov  6 20:36:49 CDlinux /usr/local/opensips_proxy/sbin/opensips[25545]:
ERROR:core:print_rr_body: too many RR
Nov  6 20:36:49 CDlinux /usr/local/opensips_proxy/sbin/opensips[25545]:
ERROR:dialog:get_routing_info: failed to print route records
Nov  6 20:36:49 CDlinux /usr/local/opensips_proxy/sbin/opensips[25545]:
ERROR:core:print_rr_body: too many RR
Nov  6 20:36:49 CDlinux /usr/local/opensips_proxy/sbin/opensips[25545]:
ERROR:dialog:get_routing_info: failed to print route records
Nov  6 20:36:49 CDlinux /usr/local/opensips_proxy/sbin/opensips[25547]:
ERROR:core:print_rr_body: too many RR
Nov  6 20:36:49 CDlinux /usr/local/opensips_proxy/sbin/opensips[25547]:
ERROR:dialog:get_routing_info: failed to print route records

i did not figure out what wrong it is going. what does RR mean,

do any module i miss to compile or load ?

thanks

Michael



On Thu, Nov 6, 2014 at 6:23 PM, Michael Leung <gbcbooksmj at gmail.com> wrote:

> Hi SamyGo
>
> My Objective is to understand how sip work
> you said that is the easiest way of how to route a call automatic , so do
> you have a better way to do that ?
>
> i want a more smart scheme to actualize dynamic routing.
>
> On Mon, Nov 3, 2014 at 12:33 AM, SamyGo <govoiper at gmail.com> wrote:
>
>> Hi Michael,
>>
>> If you HAVE to register two phones at two different OpenSIPS with
>> different domains then there are few ways to do it. I'll go with the easy
>> one.
>>
>> 1 - Static call routing
>> Define a regular expression in both OpenSIPS to dial to the other
>> OpenSIPS if the dialled destination number has certain prefix in front of
>> it and vice-versa.
>> i.e calls From 4135 at OpenSIP1.your1st-domain.com  to 94135 will route to
>> OpenSIPS2.your2nd-domain.com ;
>> At your OpenSIPS2 in the config file you've to use regexp that anything
>> having a 9 in prefix and Not from myself  then strip off prefix 9,
>> do a lookup("location") search for online user and t_relay the call to it.
>>
>> if ($tU =~ "^9*" && $fd != "OpenSIPS2.your2nd-domain.com" ) {
>> #Above detects if it has a 9 prefix and call is not coming from my own
>> users
>>
>>     strip(1);
>>
>>     if(!lookup("location") {
>> #above condition will lookup user if it is registered, if Not then return
>> below message.
>>           sl_send_reply("404", "User Offline/Not Found");
>>     } else {
>> #If user is found then relay the call to it.
>>      t_relay();
>>     }
>>
>> }
>>
>>
>> That is the easiest and static way of how you can do it. Why do you need
>> to use OpenSIPS for this? what exactly are your objectives ?
>> Hope this helped you understand a bit.
>>
>>
>> BR,
>> Sammy
>>
>> On Fri, Oct 31, 2014 at 7:13 AM, Michael Leung <gbcbooksmj at gmail.com>
>> 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 list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>
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