Hi SER users <br>
<br>
iam using ser-0.9.6 when i tried with x-lite , grandstream phones , i can make calls and recieve calls <br>
<br>
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 <br>
<br>
my ngrep messages are as follows :<br>
<br>
U <a href="http://81.21.34.34:46117">81.21.34.34:46117</a> -> <a href="http://81.21.33.35:5060">81.21.33.35:5060</a><br>
INVITE <a href="mailto:sip:32331003@81.21.33.35">sip:32331003@81.21.33.35</a> SIP/2.0.<br>
Via: SIP/2.0/UDP <a href="http://192.168.0.74:5060">192.168.0.74:5060</a>;branch=z9hG4bKH7w0kkrIoD6CyPP9.<br>
Max-Forwards: 70.<br>
User-Agent: IP-300.<br>
From: "32331001" <<a href="mailto:sip:32331001@81.21.33.35">sip:32331001@81.21.33.35</a>>;tag=ib0b3FffOYuu1C8n.<br>
To: "32331003" <<a href="mailto:sip:32331003@81.21.33.35">sip:32331003@81.21.33.35</a>>.<br>
Call-ID: <a href="mailto:HTqDhmIJuXyigmii@192.168.0.74">HTqDhmIJuXyigmii@192.168.0.74</a>.<br>
Contact: <sip:32331001@192.168.0.74:5060>.<br>
CSeq: 1 INVITE.<br>
Supported: replaces.<br>
Content-Type: application/sdp.<br>
Content-Length: 289.<br>
.<br>
v=0.<br>
o=- 32362173 86500032 IN IP4 <a href="http://192.168.0.74">192.168.0.74</a>.<br>
s=SIP CALL.<br>
c=IN IP4 <a href="http://192.168.0.74">192.168.0.74</a>.<br>
t=0 0.<br>
m=audio 8000 RTP/AVP 18 4 0 8 3 101.<br>
a=rtpmap:18 G729/8000.<br>
a=rtpmap:4 G723/8000.<br>
a=rtpmap:0 PCMU/8000.<br>
a=rtpmap:8 PCMA/8000.<br>
a=rtpmap:3 GSM/8000.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-15.<br>
<br>
#<br>
U <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> -> <a href="http://81.21.34.34:5060">81.21.34.34:5060</a><br>
SIP/2.0 407 Proxy Authentication Required.<br>
Via: SIP/2.0/UDP <a href="http://192.168.0.74:5060">192.168.0.74:5060</a>;branch=z9hG4bKH7w0kkrIoD6CyPP9;received=<a href="http://81.21.34.34">81.21.34.34</a>.<br>
From: "32331001" <<a href="mailto:sip:32331001@81.21.33.35">sip:32331001@81.21.33.35</a>>;tag=ib0b3FffOYuu1C8n.<br>
To: "32331003" <<a href="mailto:sip:32331003@81.21.33.35">sip:32331003@81.21.33.35</a>>;tag=74961b5b71b6ddce908b9155b956083f.a2b9.<br>
Call-ID: <a href="mailto:HTqDhmIJuXyigmii@192.168.0.74">HTqDhmIJuXyigmii@192.168.0.74</a>.<br>
CSeq: 1 INVITE.<br>
Proxy-Authenticate: Digest realm="<a href="http://81.21.33.35">81.21.33.35</a>", nonce="44ae1171fed2ef64959f27604e81bf11661bd0b5".<br>
Content-Length: 0.<br>
Warning: 392 <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> "Noisy feedback tells: pid=19217
req_src_ip=<a href="http://81.21.34.34">81.21.34.34</a> req_src_port=46117
in_uri=<a href="mailto:sip:32331003@81.21.33.35">sip:32331003@81.21.33.35</a> out_uri=<a href="mailto:sip:32331003@81.21.33.35">sip:32331003@81.21.33.35</a>
via_cnt==1".<br>
.<br>
<br>
#<br>
U <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> -> <a href="http://81.21.34.34:5060">81.21.34.34:5060</a><br>
SIP/2.0 410 Gone.<br>
Via: SIP/2.0/UDP <a href="http://192.168.0.74:5060">192.168.0.74:5060</a>;received=<a href="http://81.21.34.34">81.21.34.34</a>;branch=z9hG4bKZRDQYKLawgmypijq.<br>
From: <<a href="mailto:sip:32331001@81.21.33.35">sip:32331001@81.21.33.35</a>>;tag=0eZhJPf62anG4L1n.<br>
To: ravi <sip:32331003@81.21.33.35:5062>;tag=3806266128.<br>
Contact: <sip:32331003@81.21.34.34:5062>.<br>
Record-Route: <sip:32331001@81.21.33.35:5060;nat=yes;ftag=3806266128;lr=on>.<br>
Call-ID: <a href="mailto:117AD09F-713F-41BF-B61F-7AFFA70FD4C5@192.168.0.79">117AD09F-713F-41BF-B61F-7AFFA70FD4C5@192.168.0.79</a>.<br>
CSeq: 2 INVITE.<br>
Server: X-Lite release 1105x.<br>
Content-Length: 0.<br>
.<br>
<br>
<br>
What does this mean and "this phone works with Asterisk"<br>
<br>
I think i made mistake some where any body any clues please <br>
<br>
Thank You.<br>
<br>
Regards<br>