<br><br>
<div class="gmail_quote">On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner <span dir="ltr"><<a href="mailto:fred@teamforrest.com">fred@teamforrest.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>
<div></div>
<div class="h5"><br>On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote:<br><br>><br>><br>><br>> On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner <<a href="mailto:fred@teamforrest.com">fred@teamforrest.com</a>> wrote:<br>
><br>> On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote:<br>><br>> > Dear All<br>> > I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @<a href="http://192.168.0.2/" target="_blank">192.168.0.2</a> and the external sip is at @<a href="http://192.168.0.139/" target="_blank">192.168.0.139</a> . To this end , I modified my sip.conf & extensions.conf as the followings :<br>
> > Under sip.conf :<br>> > ---------------------<br>> > [general]<br>> > register => <a href="http://toronto:welcome@192.168.0.139/osaka" target="_blank">toronto:welcome@192.168.0.139/osaka</a><br>
> > [osaka]<br>> > type=friend<br>> > secret=welcome<br>> > context=osaka_incoming<br>> > host=dynamic<br>> > disallow=all<br>> > allow=alaw<br>> > [6672019]<br>> > type=friend<br>
> > host=dynamic<br>> > context=phones<br>> ><br>><br>> Try this:<br>><br>> [general]<br>> register => toronto:welcome@osaka<br>><br>> [osaka]<br>> type=friend<br>> username=toronto<br>
> authname=toronto<br>> secret=welcome<br>> context=osaka_incoming<br>> host=192.168.0.139<br>> disallow=all<br>> allow=alaw<br>><br>> Although your error shows the other server does not allow register. What is the other server?<br>
><br>> ---fred<br>> <a href="http://qxork.com/" target="_blank">http://qxork.com</a><br>><br>><br></div></div>
<div class="im">> Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ?<br>
> Thank you in advance<br>><br><br></div>
<div class="im">[general]<br>;register => toronto:welcome@osaka<br><br>[osaka]<br>type=friend<br>;username=toronto<br>;authname=toronto<br>;secret=welcome<br>context=osaka_incoming<br>host=192.168.0.139<br>disallow=all<br>
allow=alaw<br><br><br></div>
<div>
<div></div>
<div class="h5">---fred<br><a href="http://qxork.com/" target="_blank">http://qxork.com</a><br><br><br><br><br><br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
<br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div>
<div><br> </div>
<div> </div>
<div>Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server .</div>
<div>Thank you in advance</div>
<div> </div>