AsteriskWin32 does have SIP server functionality, same as the linux version.<br><br>I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work.<br>Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider...<br>
<br><div class="gmail_quote">On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi <span dir="ltr"><<a href="mailto:motamedi24@gmail.com">motamedi24@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<div class="gmail_quote"><div class="im">On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham <span dir="ltr"><<a href="mailto:dcunningham@voisonics.com" target="_blank">dcunningham@voisonics.com</a>></span> wrote:<br>
</div><div><div></div><div class="h5"><blockquote style="border-left: 1px solid rgb(204, 204, 204); 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>I downloaded & installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ?</div>
<div>Thank you</div>
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<br>_______________________________________________<br>
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IVR development, VOIP consultancy<br><a href="http://voisonics.com/">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>