[Asterisk-Users] UTStarCom F1000 SIP configuration
MF Hulber
asterisk-admin at hulber.com
Fri Jun 24 13:32:29 MST 2005
Has anyone had any luck configuring the UTStarCom F1000 with asterisk? I
get the wireless to work but the sip registration is a problem. Below is
my SIP Debug. The server is 192.168.0.80 and the phone is 192.168.0.166.
Sip.conf:
[f1000_1]
type=friend
host=dynamic
defaultip=192.168.0.166
port=5060
secret=mysecret
;auth=md5
username=f1000_1
qualify=yes
context=f1000
dtmf=rfc2833
disallow=all
allow=ulaw
mailbox=55 at mainmenu,66 at mainmenu,99 at mainmenu
canreinvite=no
nat=no
callgroup=1
pickupgroup=1
Sip Debug:
asterisk*CLI>
<-- SIP read from 192.168.0.166:5060:
REGISTER sip:192.168.0.80:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK390706295
From: <sip:f1000_1 at 192.168.0.80:5060>;tag=1979048790
To: <sip:f1000_1 at 192.168.0.80:5060>
Call-ID: 1255070303 at 192.168.0.166
CSeq: 36 REGISTER
Contact: <sip:f1000_1 at 192.168.0.166:5060>;action=proxy
Authorization: Digest username="f1000_1", realm="asterisk",
nonce="417a425e", uri="sip:192.168.0.80:5060",
response="a18b410860b77855f1952171e83e6fed", algorithm=MD5
max-forwards: 70
expires: 0
user-agent: UTSTARCOM F1000/Device ID-%ít_x˜
Content-Length: 0
--- (12 headers 0 lines)---
Using latest request as basis request
Sending to 192.168.0.166 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.166:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK390706295
From: <sip:f1000_1 at 192.168.0.80:5060>;tag=1979048790
To: <sip:f1000_1 at 192.168.0.80:5060>
Call-ID: 1255070303 at 192.168.0.166
CSeq: 36 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:f1000_1 at 192.168.0.80>
Content-Length: 0
---
Transmitting (no NAT) to 192.168.0.166:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK390706295
From: <sip:f1000_1 at 192.168.0.80:5060>;tag=1979048790
To: <sip:f1000_1 at 192.168.0.80:5060>;tag=as6abc86fb
Call-ID: 1255070303 at 192.168.0.166
CSeq: 36 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:f1000_1 at 192.168.0.80>
Content-Length: 0
---
Jun 24 16:24:56 NOTICE[9369]: chan_sip.c:9340
handle_request_register: Registration from
'<sip:f1000_1 at 192.168.0.80:5060>' failed for '192.168.0.166'
Scheduling destruction of call '1255070303 at 192.168.0.166' in 15000 ms
Destroying call '1255070303 at 192.168.0.166'
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