Hello Brian I did not specify this detail because i did not though this was important a the beginning, but the same domains are installed on freeswitch1 and freeswitch2 and mydomain.org is in fact a DNS SRV record witch point to freeswitch1 and freeswitch2. When someone on freeswitch1 wan't to call extension@domain, if it's not registered on freeswitch1 (i check using sofia_contact ) i want to forward the call to freeswitch2 and vice versa. So i think i can't do a this with a original sofia bridge but it was possible by using sip_invite_to_uri as suggested by Sergey. Now i need to secure it :-) It's probably possible to use opensips, but it's seems more complex than a pure freeswitch. On 18/09/2015 21:49, Brian West wrote:
what is wrong with:
<action application="bridge" data="sofia/profile/sip:someone@somedomain.com <mailto:sip%3Asomeone@somedomain.com>"/>
???
On Fri, Sep 18, 2015 at 12:23 PM, Tanguy <phenix@vfemail.net <mailto:phenix@vfemail.net>> wrote:
Hello Sergey
I am agree with you that my dial plan can be risky I used ACL but this can be not sufficient. I want to be able to call any internal sip extension, but i don't want that this piece dialplan to be usable reach external numbers.
I probably need to protect theses variables ( sip_to_user must be only a locally registered sip extension ) or avoid using them ( maybe using multiple conditions field for each domain ? )
I don't fully understand your initial example:
<action application="bridge" data="{sip_invite_to_uri=<sip:${destination_number}@mydomain.org <mailto:destination_number%7D@mydomain.org>>}user/reg_user@mydomain.org <mailto:reg_user@mydomain.org>"/>
What is *reg_user* for freeswitch1 ? What should i use on freeswitch2 dialplan to recognize the inbound call ?
Thanks
On 18/09/2015 07:10, Sergey Safarov wrote:
1) I recommend not use "domain" and dereviated variables. This variables may be used in bridge and other application. 2) try send invite as in example provided early.
On Thu, Sep 17, 2015, 23:22 Tanguy <phenix@vfemail.net <mailto:phenix@vfemail.net>> wrote:
I am now able to call a extension from a remote server, for those who are interested i used theses options on freeswitch2 public dialplan
<action application="set" data="call_direction=inbound"/> <action application="set" data="domain_name=${sip_to_host}" > <action application="set" data="domain=${sip_to_host}"/> <action application="transfer" data="${sip_to_user} XML ${sip_to_host}" />
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