[Asterisk-Users] Grandstream problem

[Alfie Viechweg]

Michel Belleau (malaiwah.com) wrote:

>Hi Alfie.
>
>Did you try setting up a "username=100" in your [100] context and a
>"username=101" in your [101] context?
>That should do the trick..
>
>Michel Belleau
>SERVICES INFORMATIQUES MALAIWAH.COM
>(418) 261-6412 -- http://www.malaiwah.com
>
>
>
>Alfie Viechweg a écrit :
>
>  
>
>>Can some on help me find the problem here please:
>>I'm using asterisk 1.2.0 with Grandstream GXP-2000
>>
>>This is the debugging output from asterisk:
>>
>><-- SIP read from 10.0.3.21:5060:
>>REGISTER sip:10.0.3.1 SIP/2.0
>>Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de
>>From: <sip:100 at 10.0.3.1>;tag=aea38200ad3c1539
>>To: <sip:100 at 10.0.3.1>
>>Contact: <sip:100 at 10.0.3.21>
>>Call-ID: ea87fe4398c81b7c at 10.0.3.21
>>CSeq: 10001 REGISTER
>>Expires: 3600
>>User-Agent: Grandstream GXP2000 1.0.1.9
>>Max-Forwards: 70
>>Allow:
>>INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
>>Content-Length: 0
>>
>>
>>--- (12 headers 0 lines)---
>>Using latest REGISTER request as basis request
>>Sending to 10.0.3.21 : 5060 (non-NAT)
>>Transmitting (no NAT) to 10.0.3.21:5060:
>>SIP/2.0 404 Not found
>>Via: SIP/2.0/UDP
>>10.0.3.21;branch=z9hG4bK5c77f205e9f991de;received=10.0.3.21
>>From: <sip:100 at 10.0.3.1>;tag=aea38200ad3c1539
>>To: <sip:100 at 10.0.3.1>;tag=as248942d8
>>Call-ID: ea87fe4398c81b7c at 10.0.3.21
>>CSeq: 10001 REGISTER
>>User-Agent: Asterisk PBX
>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>Max-Forwards: 70
>>Contact: <sip:100 at 10.0.3.1>
>>Content-Length: 0
>>
>>
>>---
>>Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815
>>handle_request_register: Registration from '<sip:100 at 10.0.3.1>' failed
>>for '10.0.3.21' - Username/auth name mismatch
>>Scheduling destruction of call 'ea87fe4398c81b7c at 10.0.3.21' in 15000 ms
>>Destroying call 'ea87fe4398c81b7c at 10.0.3.21'
>>
>>***************** This is the relevant parts of my sip.conf:
>>
>>[100]
>>type=friend
>>secret=test
>>qualify=yes
>>nat=no
>>host=dynamic
>>canreinvite=no
>>context=internal
>>
>>[101]
>>type=friend
>>secret=test
>>qualify=yes
>>nat=no
>>host=dynamic
>>canreinvite=no
>>context=internal
>>
>>************ This is the relevant part of my extensions.conf:
>>
>>[internal]
>>exten => 100,1,Dial(SIP/100)
>>exten => 101,1,Dial(SIP/101)
>>exten => 611,1,Echo()
>>
>>
>>
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>>
>
>  
>
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I tried adding username=xxx and that did not solve the problem.

What is the 'sip show users' command (using CLI) suppose to show in a 
properly configured server?