Zakir,<div><br></div><div>Have you checked the RFC3261?</div><div><br></div><div><div>21.4.2 401 Unauthorized</div><div>The request requires user authentication. This response is issued by</div><div>UASs and registrars, while 407 (Proxy Authentication Required) is</div>
<div>used by proxy servers.</div><div><br></div><div><br></div><br><div class="gmail_quote">2010/10/20 Zakir Mahomedy <span dir="ltr"><<a href="mailto:zmm@mayfair2000.com">zmm@mayfair2000.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div style="font-family:arial, helvetica, sans-serif;font-size:12pt"><p>Hi</p>
<p> </p>
<p>I am trying to get 2 accounts from voipblaster to talk to each other.</p>
<p>Calls withing voipblaster network is free. If I configure two sip clients with the two accounts it works fine</p>
<p>however with Asterisk I am getting SIP 401</p>
<p> </p>
<p>In my Sip.conf file I under general</p>
<p> </p>
<p>register = <a href="mailto:user%3Apassword@sip.voipblaster.com" target="_blank">user:password@sip.voipblaster.com</a></p>
<p> </p>
<p>then I have a sip peer</p>
<p> </p>
<p> </p>
<p>[FreeCall](default)<br>type= friend<br>context= incoming<br>username = kiks2010<br>secret = password<br>host= <a href="http://sip.voipblast.com" target="_blank">sip.voipblast.com</a><br>fromuser = kiks2010<br>fromdomain = <a href="http://sip.voipblast.com" target="_blank">sip.voipblast.com</a><br>
insecure=very<br>qualify=yes</p>
<p> </p>
<p>these are the sip debug logs</p>
<p> </p>
<p><font size="2">v=0<br>o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99<br>s=SIP Call<br>c=IN IP4 77.72.168.99<br>t=0 0<br>m=audio 11538 RTP/AVP 8 101<-------------></font></p>
<p><br><font size="2">--- (11 headers 9 lines) ---<br> == Using SIP RTP CoS mark 5<br>Sending to 77.72.174.128 : 5060 (NAT)<br>Using INVITE request as basis request - </font><a href="mailto:64de05c42e7b4ef2a0678f999c0edcaf@77.72.174.128" target="_blank"><font size="2">64de05c42e7b4ef2a0678f999c0edcaf@77.72.174.128</font></a><br>
<font size="2">Found peer 'FreeCall' for 'ajs2010' from <a href="http://77.72.174.128:5060" target="_blank">77.72.174.128:5060</a><br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=ptime:20</font></p>
<p><font size="2"></font> </p>
<p><font size="2"><--- Reliably Transmitting (NAT) to <a href="http://77.72.174.128:5060" target="_blank">77.72.174.128:5060</a> ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 77.72.174.128:5060;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128<br>
From: "ajs2010" <<a href="http://sip:ajs2010@sip.voipblast.com:5060" target="_blank">sip:ajs2010@sip.voipblast.com:5060</a>>;tag=330113ac4c51ef02d4ef70</font></p>
<p> </p>
<p>Any help info will be appreciated</p>
<p>thanks</p>
<p> </p><font color="#888888">
<p>Zakir</p>
<p> </p>
<p> </p></font></div></div><br>--<br>
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