Steve,<br>You're completely right!! it seems like my colleague gave me a wrong info (probably a firewall issue), i was also curious (before i read your response) so i tried this in my network and really it has nothing to do with call setup or peer authentication, sorry for the wrong info Guys! <br>
<br><div class="gmail_quote">2010/4/19 Steve Edwards <span dir="ltr"><<a href="http://asterisk.org">asterisk.org</a>@<a href="http://sedwards.com">sedwards.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Un-top-posting...<br>
<div class="im"><br>
> 2010/4/14 khalid touati <<a href="mailto:khalidtouati@gmail.com">khalidtouati@gmail.com</a>><br>
<br>
</div><div class="im">> i've connecting two pbx server successfully for several times using the following config:<br>
><br>
> register => <a href="mailto:USPBX%3Amypass@122.11.176.35">USPBX:mypass@122.11.176.35</a><br>
><br>
> [PBX1]<br>
> type=friend<br>
> host=122.11.176.35<br>
> trunk=yes<br>
> sercret=mypass<br>
> context=external<br>
> deny=<a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a><br>
> permit=<a href="http://122.11.176.35/255.255.255.240" target="_blank">122.11.176.35/255.255.255.240</a><br>
> insecure=very<br>
> allow=all<br>
> nat=yes<br>
> qualify=yes<br>
> canreinvite=no<br>
><br>
> in the other and it's the analog.<br>
><br>
> but now i can only dial from one end, and the other en d is giving me this error.<br>
><br>
> Apr 14 16:44:21 ERROR[26502]: chan_sip.c:6659 register_verify: Peer 'PBX1' is trying to register, but not<br>
> configured as host=dynamic<br>
><br>
> when dialing a fast busy signal and it sauys in the CLI: CONGESTION. any help please!!!<br>
><br>
> --<br>
> Abdullah<br>
<br>
</div><div class="im">On Mon, 19 Apr 2010, khalid touati wrote:<br>
<br>
> for people's future references: we found out that the option in DIAL<br>
> application in the extensions.conf has to be the same from both side,<br>
> the issue was India server was using "tr" while US server was using<br>
> "TWw" so we made them both using "tr" and that solved the issue, i guess<br>
> if one side is set to "trTWw" that would work regardless of the other<br>
> side but didn't try though. have a headeache-free experience with<br>
> asterisk "the future of telephony" :)!<br>
<br>
</div>Dial() options don't have any relationship to registration failures --<br>
they happen at different times.<br>
<br>
Registration failures may cause dial() failures.<br>
<br>
I don't understand the relationship between ringing, transfer and<br>
recording options and dial() returning congestion. I'd suggest<br>
investigating exactly which combination is causing congestion before<br>
concluding it is unrelated to the registration failure.<br>
<br>
--<br>
Thanks in advance,<br>
-------------------------------------------------------------------------<br>
Steve Edwards <a href="mailto:sedwards@sedwards.com">sedwards@sedwards.com</a> Voice: +1-760-468-3867 PST<br>
Newline Fax: +1-760-731-3000<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Abdullah<br>