Anton Yurchenko
2003-Mar-25 02:10 UTC
[Asterisk-Users] Continued - Turn this thread SNOM mini howto.
Hello,
I get two of my sip phones registered with asterisk,but I cant make a
call from one phone to another. The sip debug shows( see below) that
asterisk gives a 404 Not Found reply to the phone. Calling those
extensions from console works. Configs looks like:
----------
phone_name: ip-phone1
user_realname1: Anton Yurchenko
user_name1: 1001
user_host1: dg
user_action1: redirect
user_mailbox1: phila@dg
user_q1: 0.5
auth_realm1: dg
auth_user1: 1001
auth_pass1: phila
auth_valid1: 1
----------
in the sip.conf relavent section:
-----------
[1001]
type=friend
username=phila
callerid=phila
secret=phila
host=dynamic
defaultip=172.20.0.199
canreinvite=yes
mailbox=1001
-----------
and in extensions.conf
-----------
exten => _1XXX,1,Dial,sip/${EXTEN}|30|tT
-----------
SIP debug output:
------------
*CLI> sip debug
SIP Debugging Enabled
Sip read:
INVITE sip:phila.dg SIP/2.0
Via: SIP/2.0/UDP 172.20.0.199:5060;branch=z9hG4bK-u8ovolcvexaz
Max-Forwards: 70
From: "Anton Yurchenko" <sip:1001@dg>;tag=mbnks3kh3b
To: <sip:1002@dg;user=phone>
Call-ID: 3c26747b952f-lf1v5z385h2h@172.20.0.199
CSeq: 1 INVITE
Route: <sip:1002@dg;user=phone>
Contact: <sip:1001@172.20.0.199:5060>
User-Agent: snom Version 1.15u
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSA
GE
Supported: timer, 100rel, replaces
Session-Expires: 7200
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 16533 16533 IN IP4 172.20.0.199
s=SIP Call
c=IN IP4 172.20.0.199
t=0 0
m=audio 10002 RTP/AVP 0 8 3 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
17 headers, 12 lines
Interface is eth0
IP Address is 172.20.0.50
Using latest request as basis request
Sending to 172.20.0.199 : 5060 (non-NAT)
Capabilities: us - 14, them - 270, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.20.0.199:5060;branch=z9hG4bK-u8ovolcvexaz
From: "Anton Yurchenko" <sip:1001@dg>;tag=mbnks3kh3b
To: <sip:1002@dg;user=phone>;tag=32283a32
Call-ID: 3c26747b952f-lf1v5z385h2h@172.20.0.199
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:1002@172.20.0.50>
Proxy-Authenticate: Digest realm="asterisk",
nonce="50583749"
Content-Length: 0
to 172.20.0.199:5060
Sip read:
ACK sip:phila.dg SIP/2.0
Via: SIP/2.0/UDP 172.20.0.199:5060;branch=z9hG4bK-u8ovolcvexaz
Max-Forwards: 70
From: "Anton Yurchenko" <sip:1001@dg>;tag=mbnks3kh3b
To: <sip:1002@dg;user=phone>;tag=32283a32
Call-ID: 3c26747b952f-lf1v5z385h2h@172.20.0.199
CSeq: 1 ACK
Route: <sip:1002@dg;user=phone>
Contact: <sip:1001@172.20.0.199:5060>
Content-Length: 0
10 headers, 0 lines
Sip read:
INVITE sip:phila.dg SIP/2.0
Via: SIP/2.0/UDP 172.20.0.199:5060;branch=z9hG4bK-zzw8oxu50m0w
Max-Forwards: 70
From: "Anton Yurchenko" <sip:1001@dg>;tag=mbnks3kh3b
To: <sip:1002@dg;user=phone>
Call-ID: 3c26747b952f-lf1v5z385h2h@172.20.0.199
CSeq: 2 INVITE
Route: <sip:1002@dg;user=phone>
Contact: <sip:1001@172.20.0.199:5060>
User-Agent: snom Version 1.15u
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSA
GE
Supported: timer, 100rel, replaces
Session-Expires: 7200
Content-Type: application/sdp
Content-Length: 263
Proxy-Authorization: Digest
username="1001",realm="asterisk",nonce="50583749",ur
i="sip:",response="30bbc956ca4f62151355a39eb2015298",algorithm=md5
v=0
o=root 16533 16533 IN IP4 172.20.0.199
s=SIP Call
c=IN IP4 172.20.0.199
t=0 0
m=audio 10002 RTP/AVP 0 8 3 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
18 headers, 12 lines
Using latest request as basis request
Sending to 172.20.0.199 : 5060 (non-NAT)
Capabilities: us - 14, them - 270, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for phila.dg in local
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.20.0.199:5060;branch=z9hG4bK-zzw8oxu50m0w
From: "Anton Yurchenko" <sip:1001@dg>;tag=mbnks3kh3b
To: <sip:1002@dg;user=phone>;tag=32283a32
Call-ID: 3c26747b952f-lf1v5z385h2h@172.20.0.199
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:1002@172.20.0.50>
Content-Length: 0
to 172.20.0.199:5060
Sip read:
ACK sip:phila.dg SIP/2.0
Via: SIP/2.0/UDP 172.20.0.199:5060;branch=z9hG4bK-zzw8oxu50m0w
Max-Forwards: 70
From: "Anton Yurchenko" <sip:1001@dg>;tag=mbnks3kh3b
To: <sip:1002@dg;user=phone>;tag=32283a32
Call-ID: 3c26747b952f-lf1v5z385h2h@172.20.0.199
CSeq: 2 ACK
Route: <sip:1002@dg;user=phone>
Contact: <sip:1001@172.20.0.199:5060>
Content-Length: 0
------------
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation