<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:arial, helvetica, sans-serif;font-size:12pt"><P>Hi</P>
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<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>
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<P>In my Sip.conf file I under general</P>
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<P>register = user:password@sip.voipblaster.com</P>
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<P>then I have a sip peer</P>
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<P>[FreeCall](default)<BR>type= friend<BR>context= incoming<BR>username = kiks2010<BR>secret = password<BR>host= sip.voipblast.com<BR>fromuser = kiks2010<BR>fromdomain = sip.voipblast.com<BR>insecure=very<BR>qualify=yes</P>
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<P>these are the sip debug logs</P>
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<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"><FONT size=2>64de05c42e7b4ef2a0678f999c0edcaf@77.72.174.128</FONT></A><BR><FONT size=2>Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060<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 77.72.174.128:5060 ---><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" <sip:ajs2010@sip.voipblast.com:5060>;tag=330113ac4c51ef02d4ef70</FONT></P>
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<P>Any help info will be appreciated</P>
<P>thanks</P>
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<P>Zakir</P>
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