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<div class="gmail_quote">On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham <span dir="ltr"><<a href="mailto:dcunningham@voisonics.com">dcunningham@voisonics.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hadi,<br><br>You could use Asterisk as a sip server, it's installable on Windows.<br><br>Using "sip set debug on" might help you with the "Host '192.168.0.139' does not implement 'REGISTER'" problem.
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<div class="gmail_quote">On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi <span dir="ltr"><<a href="mailto:motamedi24@gmail.com" target="_blank">motamedi24@gmail.com</a>></span> wrote:<br>
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<div class="gmail_quote">On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner <span dir="ltr"><<a href="mailto:fred@teamforrest.com" target="_blank">fred@teamforrest.com</a>></span> wrote:<br>
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<div><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" target="_blank">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>> 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>[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>
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<div>---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>
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<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><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>
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<br></div></div><font color="#888888">-- <br>David Cunningham<br>Voisonics<br>IVR development, VOIP consultancy<br><a href="http://voisonics.com/" target="_blank">http://voisonics.com/</a><br>US toll-free: +1 888 842 2720<br>
UK: +44 (0) 20 3411 5024<br>Australia: +61 (0) 2 9037 2180<br><br></font><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
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<div>With many thanks , please let me to ask you if I rely upon on my Asterisk 1.4 installation on my CentOS 5.0 and want to have this "external sip client" on my CentOS server as well so what will be the solution ? The one you told me was for the Laboratory test when the Asterisk on CentOS calls sip client on MS Windows but what will be the solution if the Asterisk on CentOS calls sip client on the same CentOS ? Is there a Voip application on the CentOS that can resemble this "external sip client" ?</div>
<div>Thank you</div>
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