[asterisk-users] SIP 401

Danny Dias ing.diasdanny at gmail.com
Wed Oct 20 12:08:16 CDT 2010


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 <zmm at mayfair2000.com>

> 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:password at sip.voipblaster.com<user%3Apassword at sip.voipblaster.com>
>
>
>
> 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 -
> 64de05c42e7b4ef2a0678f999c0edcaf at 77.72.174.128
> 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:ajs2010 at sip.voipblast.com:5060
> >;tag=330113ac4c51ef02d4ef70
>
>
>
> Any help info will be appreciated
>
> thanks
>
>
>
> Zakir
>
>
>
>
>
> --
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