Hi,<br><br>I would like to achive such thing:<br><br>Asterisk is gateway between two sip domain domain.a and domain.b.<br>There are two users registered in domain A: bob1 and bob2.<br>When they are making outbound calls the calls should go through the asterisk gateway.
<br>Asterisk has two users registered in domain B sally1 and sally2 (with the usage of 'register =>' command in sip.conf).<br>when there is a call initiated by <a href="mailto:bob1@domain.a">bob1@domain.a</a> I would like asterisk to pass it further as the user
<a href="mailto:sally1@domain.b">sally1@domain.b</a> and similary when call is initiated by <a href="mailto:bob2@domain.b">bob2@domain.b</a> <br>to pass it as <a href="mailto:sally2@domain.b">sally2@domain.b</a>. So can I somehow check who is making the call from domain A and according to this criteria make a decision which user from domain B will be assigned in asterisk to pass it further??
<br><br>Thank you for any help.<br><br>Bests regards<br><br><div><span class="gmail_quote">On 11/4/07, <b class="gmail_sendername">Dovid B</b> <<a href="mailto:asteriskusers@dovid.net">asteriskusers@dovid.net</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff">
<div><font face="Arial" size="2">If you just want all users to register with "Domain
A" and call over a single SIP trunk to "Domain B" that would be fairly simple.
You have all the phones register on "Domain A" and set that all calls got
through trunk X to "Domain B". If you want "Domain B" to call a specific phone
connected to "Domain A" then it might be a bit trickier. </font></div>
<blockquote style="border-left: 2px solid rgb(0, 0, 0); padding-right: 0px; padding-left: 5px; margin-left: 5px; margin-right: 0px;"><div><span class="e" id="q_11607d25e6dc571a_1">
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">----- Original Message ----- </div>
<div style="background: rgb(228, 228, 228) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">
<b>From:</b>
<a title="tzieleniewski@gmail.com" href="mailto:tzieleniewski@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Tomasz
Zieleniewski</a> </div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>To:</b> <a title="asterisk-users@lists.digium.com" href="mailto:asterisk-users@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-users@lists.digium.com</a>
</div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Sent:</b> Friday, November 02, 2007 5:25
PM</div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Subject:</b> [asterisk-users] Asterisk as a
SIP to SIP Gateway</div>
<div><br></div>Hi,<br><br>It is my second time when I try to use asterisk
:)<br>I am starting with the following issue.<br>I want asterisk to behave as
a gateway between two sip networks.<br><br>My architecture is the
following:<br><br>SIP proxy (registrar - domain A)
--------------- Asterisk Gateway ------------------------ SIP
proxy (registrar - domain B)
<br>
(number of UAC registered in domain B)<br><br>Asterisk form the point of view
of the domain A and users registered in A is outbound proxy and from the point
of view of domain B is a set of SIP clients. <br>Is it possible to configure
asterisk in such way that for a particular user from domain A who calls
through asterisk <br>the SIP signaling will be passed through some pointed
user (asterisk user logged in domain B).<br>It is easy to achieve that in the
other direction. when there is a connection from domain B to a particular user
which was registered<br>by asterisk. One can forward such connection to some
user registered in sip proxy in domain A. <br>Please point me how can make
such a mapping between the calling user from domain A and a user in asterisk
who is registered in domain B??<br><br>Thank You in
advance.<br>Regards<br><br>Tomasz<br><br>
</span></div><p>
</p><hr>
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