[Asterisk-Users] Grandstream problem
Michel Belleau (malaiwah.com)
michel.belleau at malaiwah.com
Thu Nov 24 16:47:50 MST 2005
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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