[Serusers] Cannot make Out going calls

[ravi reddy]

Hi SER users

iam using ser-0.9.6 when i tried with x-lite , grandstream phones , i can
make calls and recieve calls

now iam using one new phone it is  'ip-300' SIP phone i can register taht
phone and i can make calls to that 'ip-300' but when I tried to make calls
from that phone it is not success

my ngrep messages are as follows :

U 81.21.34.34:46117 -> 81.21.33.35:5060
INVITE sip:32331003 at 81.21.33.35 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bKH7w0kkrIoD6CyPP9.
Max-Forwards: 70.
User-Agent: IP-300.
From: "32331001" <sip:32331001 at 81.21.33.35>;tag=ib0b3FffOYuu1C8n.
To: "32331003" <sip:32331003 at 81.21.33.35>.
Call-ID: HTqDhmIJuXyigmii at 192.168.0.74.
Contact: <sip:32331001 at 192.168.0.74:5060>.
CSeq: 1 INVITE.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 289.
.
v=0.
o=- 32362173 86500032 IN IP4 192.168.0.74.
s=SIP CALL.
c=IN IP4 192.168.0.74.
t=0 0.
m=audio 8000 RTP/AVP 18 4 0 8 3 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 81.21.33.35:5060 -> 81.21.34.34:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bKH7w0kkrIoD6CyPP9;received=
81.21.34.34.
From: "32331001" <sip:32331001 at 81.21.33.35>;tag=ib0b3FffOYuu1C8n.
To: "32331003" <sip:32331003 at 81.21.33.35>;tag=
74961b5b71b6ddce908b9155b956083f.a2b9.
Call-ID: HTqDhmIJuXyigmii at 192.168.0.74.
CSeq: 1 INVITE.
Proxy-Authenticate: Digest realm="81.21.33.35",
nonce="44ae1171fed2ef64959f27604e81bf11661bd0b5".
Content-Length: 0.
Warning: 392 81.21.33.35:5060 "Noisy feedback tells:  pid=19217 req_src_ip=
81.21.34.34 req_src_port=46117 in_uri=sip:32331003 at 81.21.33.35 out_uri=
sip:32331003 at 81.21.33.35 via_cnt==1".
.

#
U 81.21.33.35:5060 -> 81.21.34.34:5060
SIP/2.0 410 Gone.
Via: SIP/2.0/UDP 192.168.0.74:5060;received=81.21.34.34
;branch=z9hG4bKZRDQYKLawgmypijq.
From: <sip:32331001 at 81.21.33.35>;tag=0eZhJPf62anG4L1n.
To: ravi <sip:32331003 at 81.21.33.35:5062>;tag=3806266128.
Contact: <sip:32331003 at 81.21.34.34:5062>.
Record-Route: <sip:32331001 at 81.21.33.35:5060;nat=yes;ftag=3806266128;lr=on>.
Call-ID: 117AD09F-713F-41BF-B61F-7AFFA70FD4C5 at 192.168.0.79.
CSeq: 2 INVITE.
Server: X-Lite release 1105x.
Content-Length: 0.
.


What does this mean and "this phone works with Asterisk"

I think i made mistake some where any body any clues please

                                   Thank You.

Regards
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