[Asterisk-Users] Grandstream problem

Alfie Viechweg alfie at syncompute.net
Thu Nov 24 16:45:06 MST 2005


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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