<div><div class="gmail_quote">On Tue, May 10, 2011 at 1:59 AM, bob yang <span dir="ltr"><<a href="mailto:learnin9@gmail.com">learnin9@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
hi,Andrew Latham:<br> Thanks for your response! sorry my poor written english!<br> OK,I write my testing process......<br><span style="color:rgb(255, 0, 0)"> </span><b style="color:rgb(255, 0, 0)"> first step:</b><br>
(1): add the SIP trunk and try to register the remote SIP softswitch(another asterisk system.<img style="margin:0pt 0.2ex;vertical-align:middle">the remote SIP port is 5060)<br>
(2): add the dial plan, _1XX to dial the other system's sip phone.<br> (3): add a sip phone and make it to register the sip system(A)<br> (4): call 100.the two call can be connted successfully.<br> please see the attachment asterisk_5060.txt<br>
<b> <span style="color:rgb(255, 0, 0)"> Second step:</span></b><br> (1): Make the remote SIP system port 5080<br>
(2): Modify the SIP trunk port 5080 and "Apply changes". (Yes, it can
register the remote sip sysyem and the remote port is 5080)<br>
(3): call 100, I found the SIP system send "480" error messages<br> Please see the attachment asterisk_5080.txt<br><br> Just modified the SIP register port not 5060, the call was failed......<br> In the asterisk_5080.txt file , I found the error messages, maybe it's the reason! but i do not know the reason.....<br>
<span style="color:rgb(0, 0, 153)"> [May 10 13:37:07] WARNING[14650]: chan_sip.c:3094 create_addr: No such host: trunk_1</span><br style="color:rgb(0, 0, 153)"><span style="color:rgb(0, 0, 153)"> [May 10 13:37:07] DEBUG[14650]: chan_sip.c:17346 sip_request_call: Cant create SIP call - target device not registered</span><br>
Thank you!<div><div></div><div class="h5"><br> <br><br><div class="gmail_quote">2011/5/8 Erin Spiceland <span dir="ltr"><<a href="mailto:espiceland@digium.com" target="_blank">espiceland@digium.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>> I use the GUI on over 60 servers. I am more than willing to help test.<br>
> I have servers with up to 3000 users and use every feature. My<br>
> question is how do I help, is there a place you want feedback?<br>
<br>
</div>"Works for me" messages can go to the list. If you find bugs, please enter them in the issue tracker at <a href="http://issues.asterisk.org" target="_blank">http://issues.asterisk.org</a>. Thanks!<br>
<font color="#888888"><br>
Erin</font></blockquote></div></div></div></blockquote></div><br clear="all"><meta http-equiv="content-type" content="text/html; charset=utf-8">Bob, thanks for the input...<div><br></div><div>If / When you have a config issue that does not work in Asterisk, report it as a configuration conflict. For example [1] is a conflict of the GUI with its own settings. Add an issue and what you think is supposed to happen. For a SIP port change you can show it working in a normal Asterisk installation first then add the GUI.</div>
<div><br></div><div>1. <a href="https://issues.asterisk.org/view.php?id=18181">https://issues.asterisk.org/view.php?id=18181</a></div></div><div><br>-- <br>~~~ Andrew "lathama" Latham <a href="mailto:lathama@gmail.com" target="_blank">lathama@gmail.com</a> ~~~<br>
</div>