By the way, Could you please make a "better picture" of your work?
try using insecure=invite,port, that's the key! by the way, try to use IPs rather than domain names. And check here also: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf register => user[:secret[:authuse...@host[:port][/extension] 2010/10/20 Danny Dias <[email protected]> > Zakir, > > Have you checked the RFC3261? > > 21.4.2 401 Unauthorized > The request requires user authentication. This response is issued by > UASs and registrars, while 407 (Proxy Authentication Required) is > used by proxy servers. > > > > 2010/10/20 Zakir Mahomedy <[email protected]> > >> Hi >> >> >> >> I am trying to get 2 accounts from voipblaster to talk to each other. >> >> Calls withing voipblaster network is free. If I configure two sip >> clients with the two accounts it works fine >> >> however with Asterisk I am getting SIP 401 >> >> >> >> In my Sip.conf file I under general >> >> >> >> register = >> user:[email protected]<user%[email protected]> >> >> >> >> then I have a sip peer >> >> >> >> >> >> [FreeCall](default) >> type= friend >> context= incoming >> username = kiks2010 >> secret = password >> host= sip.voipblast.com >> fromuser = kiks2010 >> fromdomain = sip.voipblast.com >> insecure=very >> qualify=yes >> >> >> >> these are the sip debug logs >> >> >> >> v=0 >> o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 >> s=SIP Call >> c=IN IP4 77.72.168.99 >> t=0 0 >> m=audio 11538 RTP/AVP 8 101<-------------> >> >> >> --- (11 headers 9 lines) --- >> == Using SIP RTP CoS mark 5 >> Sending to 77.72.174.128 : 5060 (NAT) >> Using INVITE request as basis request - >> [email protected] >> Found peer 'FreeCall' for 'ajs2010' from 77.72.174.128:5060 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=ptime:20 >> >> >> >> <--- Reliably Transmitting (NAT) to 77.72.174.128:5060 ---> >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 77.72.174.128:5060 >> ;branch=z9hG4bK6ff0e241f3fd4d0b9c137d616de1fe1f;received=77.72.174.128 >> From: "ajs2010" <sip:[email protected]:5060 >> >;tag=330113ac4c51ef02d4ef70 >> >> >> >> Any help info will be appreciated >> >> thanks >> >> >> >> Zakir >> >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >
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