<div>Hey,</div><div><br></div>The callee server is complaining too loud "<span class="Apple-style-span" style="font-family: 'Courier New'; font-size: 13px; ">Call from '2765' to </span><span class="Apple-style-span" style="font-family: 'Courier New'; "><span class="Apple-style-span" style="background-color: rgb(255, 0, 0);">extension '<b>1166:password</b></span>' rejected because <u>extension not found</u>.</span>" <div>
Try changing the Dial string as DIAL(SIP/<span class="Apple-style-span" style="font-family: 'Courier New'; font-size: 13px; ">asterisk-callee/${EXTEN}</span>) or w/e extension you require in place of ${EXTEN}</div>
<div>Let me know what changes.</div><div><br></div><div>Also this is a good read: <a href="http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/">http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/</a></div><div><br>
</div><div><div>Wed, Sep 14, 2011 at 12:37 PM, Lee, John (Sydney) <span dir="ltr"><<a href="mailto:John.Lee@compuware.com">John.Lee@compuware.com</a>></span> wrote:<div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div></div><div class="h5"><div lang="EN-AU" link="blue" vlink="purple"><div><p><span style="font-size:10.0pt;font-family:"Courier New"">I was trying to do a SIP call between two Asterisk servers (1.4.21.2)<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New""><u></u> <u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">1) On the caller server, I coded the following in extensions.conf<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">Dial(SIP/1166:password@asterisk-callee);<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New""><u></u> <u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">2) On the callee server, I coded the following in sip.conf<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">[1166]<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">type=friend ; Friends place calls and receive calls<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">context=incoming ; Context for incoming calls from this user<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">host=dynamic ; This peer register with us<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">qualify=yes ; Monitor latency between Asterisk server and phone<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">call-limit=99<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">username=1166 ; Username to use in INVITE until peer registers<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">secret=password ; Normally you do NOT need to set this parameter<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">mailbox=1166@default ; mailbox 5100 in voicemail context .default.<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">callgroup=1<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">pickupgroup=1<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New""><u></u> <u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">The call was unsuccessful as follows.<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New""> <u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">1) On the caller machine, this is what we got from the console<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New""> -- Executing [1166@incoming:1] Dial("SIP/1166-09d81668", "SIP/1166:password@asterisk-callee") in new stack<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New""> -- Called 1166:password@asterisk-callee<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New""> -- SIP/asterisk-callee is circuit-busy<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New""> == Everyone is busy/congested at this time (1:0/1/0)<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New""><u></u> <u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New"">2) On the callee machine, this is what we got from the console,<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">[Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite: Call from '2765' to extension '1166:password' rejected because extension not found.<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New""><u></u> <u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">However, I found out that if I remove “secret=..” from the SIP entry and call without the password, then I will be able to call.<u></u><u></u></span></p>
<p><span style="font-size:10.0pt;font-family:"Courier New""><u></u> <u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New"">Any thoughts?<u></u><u></u></span></p><p><span style="font-size:10.0pt;font-family:"Courier New""><u></u> <u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p></div></div></div></div><br>--<br>
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