Unfortunately can't apply this patch to asterisk-1.6.1-svn-r161639.<br>
<br>
It is rejected with an error "Hunk #1 FAILED at 3953."<br><br><div class="gmail_quote">2008/12/9 Matthew Nicholson <span dir="ltr"><<a href="mailto:mnicholson@digium.com">mnicholson@digium.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;">There is at least one problem that I know of. Try the patch from bug<br>
12006 (<a href="http://bugs.digium.com/view.php?id=12006" target="_blank">http://bugs.digium.com/view.php?id=12006</a>).<br>
<div><div></div><div class="Wj3C7c"><br>
On Tue, 2008-12-09 at 15:51 +0000, Chris Maciejewski wrote:<br>
> Hi,<br>
><br>
> I am trying to force Asterisk (SVN-branch-1.6.1-r161639) to send all<br>
> SIP signalling via a proxy server.<br>
><br>
> Unfortunately when I put in sip.conf:<br>
> [general]<br>
> ...<br>
> outboundproxy=proxy.domain:5060<br>
><br>
> and try to Dial(SIP/<a href="mailto:enum-test@sip.nemox.net">enum-test@sip.nemox.net</a>), I am getting the<br>
> following error in the console:<br>
><br>
> -- Executing [43780004711@dialSIP:3]<br>
> Dial("SIP/dev-sip.tele500.com-08204d10",<br>
> "SIP/<a href="mailto:enum-test@sip.nemox.net">enum-test@sip.nemox.net</a>") in new stack<br>
> == Using SIP RTP CoS mark 5<br>
> [Dec 9 15:39:35] ERROR[2344]: chan_sip.c:19423 handle_request_do: We<br>
> could NOT get the channel lock for SIP/sip.nemox.net-08210dc8!<br>
> [Dec 9 15:39:35] ERROR[2344]: chan_sip.c:19424 handle_request_do: SIP<br>
> transaction failed: <a href="mailto:365c9c8209a3163523bd79782dc9d208@78.105.1.129">365c9c8209a3163523bd79782dc9d208@78.105.1.129</a><br>
> -- Got SIP response 503 "Server error" back from <a href="http://0.0.0.0" target="_blank">0.0.0.0</a><br>
> -- Called <a href="mailto:enum-test@sip.nemox.net">enum-test@sip.nemox.net</a><br>
> -- SIP/sip.nemox.net-08210dc8 is circuit-busy<br>
><br>
> When I use IP address instead of a domain name:<br>
> [general]<br>
> ...<br>
> outboundproxy=proxy_IP_address:5060<br>
><br>
> There is an error as below:<br>
><br>
> -- Executing [43780004711@dialSIP:3]<br>
> Dial("SIP/dev-sip.tele500.com-b7208810",<br>
> "SIP/<a href="mailto:enum-test@sip.nemox.net">enum-test@sip.nemox.net</a>") in new stack<br>
> == Using SIP RTP CoS mark 5<br>
> -- Called <a href="mailto:enum-test@sip.nemox.net">enum-test@sip.nemox.net</a><br>
> -- Got SIP response 482 "Loop Detected" back from <a href="http://0.0.0.0" target="_blank">0.0.0.0</a><br>
><br>
> In both cases no SIP packets are leaving Asterisk.<br>
><br>
> Am I missing something, or there is a problem with outboundproxy in<br>
> "general" section of sip.conf file.<br>
><br>
> Kind regards,<br>
><br>
> Chris<br>
><br>
</div></div>> _______________________________________________<br>
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--<br>
Matthew Nicholson<br>
Digium<br>
</blockquote></div><br>