<div>Well, unfortunately i did not dig much into "why/how it worked" with openvpn, but it did work for me with default setup.I think you may need to set constant ports instead of random ports.</div>
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<div>Thanks,</div>
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<div>Vivek<br><br> </div>
<div><span class="gmail_quote">On 11/9/07, <b class="gmail_sendername">bilal ghayyad</b> <<a href="mailto:bilmar_gh@yahoo.com">bilmar_gh@yahoo.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi Friends;<br><br>Actually I would appreciate if Vivek can advise if the<br>VPN resolved the RTP packets in the SIP Trunk between
<br>Asterisk and another softswitch? In other words,<br>openvpn helpful in NAT cases in what exactly? As<br>without VPN, I was able to establish a call but<br>without voice or with complete noise (nothing<br>understood) :) - So if NAT resolve this issue for the
<br>SIP Trunk, then I can proceed forward, as really now I<br>do not have any other attempt to try.<br><br>From the other side, I think that baji is talking<br>about something else than the IP Trunk, he is talking<br>about outbound (which is related to using an
<br>application to run an outside call, which is used<br>usually in campaign in contact centers and so on), I<br>think nthis case differs that placing a calls via IP<br>Trunk or even outside call but the caller who will do
<br>it (and not the application).<br><br>Lastly, Mr. Amit helped me when he gave me a<br>configuration to be done for the SIP Trunk, as in his<br>method, I did not register on the softswitch, I send<br>directly, and the connectioned succeed, but as I said:
<br>with complete voice (actually nothing understood, i<br>feel it is complete RTP situation), the test was by<br>letting Asterisk behind NAT (private IP) and sending<br>to a softswitch in anther country has a public IP<br>
address. Is it NAT issue, so VPN can resolve?<br><br>Note: anyone knows if h323 works better in the IP<br>trunk?<br><br>Regards<br>Bilal<br><br>----------------------------------<br>yeah i found openvpn helpful in NAT cases.
<br><br>-Vivek<br><br><br>On 11/6/07, Baji Panchumarti<br><<a href="mailto:baji.panchumarti@gmail.com">baji.panchumarti@gmail.com</a>> wrote:<br>><br>> after a copious loss of follicles :-), I finally got<br>outbound
<br>working.<br>><br>> Basically the channel statement in the call file<br>needs to have the<br>> number to be called. For eg., in test.call format<br>the statement<br>> as follows :<br>><br>> Channel: SIP/3012345678@<your-sip-provider>
<br>><br>> And there is no need for a DIAL statement in<br>extensions.conf<br>> unless you need to dial an additional number /<br>extension.<br>><br>> Then in sip.conf you need a para that matches<br><your-sip-provider>
<br>> with the relevant auth info.<br>><br>> These two wiki pages, they were very helpful in<br>figuring out a<br>> solution to the problem :<br>><br>><br><a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out">
http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out</a><br>><br>><br>><br><br><a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message">http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message
</a><br>><br>> hth,<br>><br>> -baji.<br>><br>> --<br>><br>> On Oct 30, 2007 8:43 AM, Gabriel Natale wrote:<br>><br>> > I have the same problem.<br>> ><br>> > I trying with more 4 SIP providers, the account is
<br>registering,<br>receive<br>> > inboud calls, but can`t make outbound calls for<br>"congestion".<br>> ><br>> > Can be the out call id the problem?<br>> ><br>> > Thanks<br>> > Gabriel
<br><br><br>__________________________________________________<br>Do You Yahoo!?<br>Tired of spam? Yahoo! Mail has the best spam protection around<br><a href="http://mail.yahoo.com">http://mail.yahoo.com</a><br></blockquote>
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