Hi all,
I am struggling to make my asterisk server work. The problem is I can not
place a call from a phone outside, but I can call out from a phone in the
local network where the asterisk server sits.
I turn the debug on, and the log are shown below. I can see "REGISTER"
method is OK. ( SIP/2.0 200 OK) But Later, in the "INVITE" method,
the
SIP addresses become something like:
From: "10916"
<sip:219.xx.xx.9@60.xx.xx.164>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:192.168.1.2@60.xx.xx.164>
which have IP addresses in the part before @. Then I got a "SIP/2.0 404 Not
Found" and the call failed. I am very confused by this type of SIP
addresses. Are they valid? If it is a problem, how can I fix it? Thanks in
advance.
Cheers,
Dan
Here is my setup.
asterisk server DMZ of an ADSL router, public IP=60.xx.xx.164, local IP =
192.168.1.2
phone 10916: behind a router, public IP = 219.xx.xx.9
phone 10920: behind a router, public IP = 218.xx.xx.24
***************************************************
Sip read:
REGISTER sip:60.xx.xx.164 SIP/2.0
Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bKVfTeS1TbJ
Max-Forwards: 70
User-Agent: PA168S
From: "10916" <sip:10916@60.xx.xx.164>;tag=ks8tHkBudRSI7Ydz
To: "10916" <sip:10916@60.xx.xx.164>
Call-ID: l50TKpxGbtLGYIvi@218.xx.xx.24
CSeq: 25463 REGISTER
Contact: <sip:10916@219.xx.xx.9:5060>
Expires: 60
Content-Length: 0
11 headers, 0 lines
Using latest request as basis request
Sending to 219.xx.xx.9 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
219.xx.xx.9:5060;branch=z9hG4bKVfTeS1TbJ;received=219.xx.xx.9;rport=5060
From: "10916" <sip:10916@60.xx.xx.164>;tag=ks8tHkBudRSI7Ydz
To: "10916" <sip:10916@60.xx.xx.164>;tag=as4016e46b
Call-ID: l50TKpxGbtLGYIvi@218.xx.xx.24
CSeq: 25463 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:10916@60.xx.xx.164>
Content-Length: 0
to 219.xx.xx.9:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
219.xx.xx.9:5060;branch=z9hG4bKVfTeS1TbJ;received=219.xx.xx.9;rport=5060
From: "10916" <sip:10916@60.xx.xx.164>;tag=ks8tHkBudRSI7Ydz
To: "10916" <sip:10916@60.xx.xx.164>;tag=as4016e46b
Call-ID: l50TKpxGbtLGYIvi@218.xx.xx.24
CSeq: 25463 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:10916@60.xx.xx.164>
WWW-Authenticate: Digest realm="asterisk", nonce="44ae239c"
Content-Length: 0
to 219.xx.xx.9:5060
Scheduling destruction of call 'l50TKpxGbtLGYIvi@218.xx.xx.24' in 15000
ms
Sip read:
REGISTER sip:60.xx.xx.164 SIP/2.0
Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bKzsNF9bGEp
Max-Forwards: 70
User-Agent: PA168S
From: "10916" <sip:10916@60.xx.xx.164>;tag=uL0iKjrTXppeLeoq
To: "10916" <sip:10916@60.xx.xx.164>
Call-ID: l50TKpxGbtLGYIvi@218.xx.xx.24
CSeq: 25464 REGISTER
Contact: <sip:10916@219.xx.xx.9:5060>
Expires: 60
Authorization: Digest username="10916", realm="asterisk",
nonce="44ae239c",
uri="sip:60.xx.xx.164",
response="344518534345eb0f6d60e30c13b81c35"
Content-Length: 0
12 headers, 0 lines
Using latest request as basis request
Sending to 219.xx.xx.9 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
219.xx.xx.9:5060;branch=z9hG4bKzsNF9bGEp;received=219.xx.xx.9;rport=5060
From: "10916" <sip:10916@60.xx.xx.164>;tag=uL0iKjrTXppeLeoq
To: "10916" <sip:10916@60.xx.xx.164>;tag=as4016e46b
Call-ID: l50TKpxGbtLGYIvi@218.xx.xx.24
CSeq: 25464 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:10916@60.xx.xx.164>
Content-Length: 0
to 219.xx.xx.9:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
219.xx.xx.9:5060;branch=z9hG4bKzsNF9bGEp;received=219.xx.xx.9;rport=5060
From: "10916" <sip:10916@60.xx.xx.164>;tag=uL0iKjrTXppeLeoq
To: "10916" <sip:10916@60.xx.xx.164>;tag=as4016e46b
Call-ID: l50TKpxGbtLGYIvi@218.xx.xx.24
CSeq: 25464 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:10916@219.xx.xx.9:5060>;expires=60
Date: Tue, 01 Feb 2005 03:08:46 GMT
Content-Length: 0
to 219.xx.xx.9:5060
Scheduling destruction of call 'l50TKpxGbtLGYIvi@218.xx.xx.24' in 15000
ms
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:219.xx.xx.9 SIP/2.0
Via: SIP/2.0/UDP 60.xx.xx.164:5060;branch=z9hG4bK6bc95bb5
From: "asterisk" <sip:asterisk@60.xx.xx.164>;tag=as4e42ecfb
To: <sip:219.xx.xx.9>
Contact: <sip:asterisk@60.xx.xx.164>
Call-ID: 3676dea078e07a6f1e6b17084473bdea@60.xx.xx.164
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 01 Feb 2005 03:08:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 219.xx.xx.9:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6bc95bb5
Call-ID: 3676dea078e07a6f1e6b17084473bdea@60.xx.xx.164
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as4e42ecfb
To: <sip:219.xx.xx.9>;tag=zIQDT3uvUSCwzF2y
Contact: <sip:219.xx.xx.9@219.xx.xx.9:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO,
PRACK, UPDATE
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 212
v=0
o=10916 86824506 47524594 IN IP4 219.xx.xx.9
s=SIP CALL
c=IN IP4 219.xx.xx.9
t=0 0
m=audio 5004 RTP/AVP 18 4 0 8
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
14 headers, 10 lines
Destroying call '3676dea078e07a6f1e6b17084473bdea@60.xx.xx.164'
Sip read:
INVITE sip:192.168.1.2@60.xx.xx.164 SIP/2.0
Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bK8HxS8oAgT
Max-Forwards: 70
User-Agent: PA168S
From: "10916"
<sip:219.xx.xx.9@60.xx.xx.164>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:192.168.1.2@60.xx.xx.164>
Call-ID: BkmCUMXIAe0otg6w@219.xx.xx.9
Contact: <sip:219.xx.xx.9@219.xx.xx.9:5060>
CSeq: 1 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO,
PRACK, UPDATE
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 212
v=0
o=10916 77859998 41941006 IN IP4 219.xx.xx.9
s=SIP CALL
c=IN IP4 219.xx.xx.9
t=0 0
m=audio 5004 RTP/AVP 18 4 0 8
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
13 headers, 10 lines
Using latest request as basis request
Sending to 219.xx.xx.9 : 5060 (NAT)
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 219.xx.xx.9:5004
Found description format G729
Found description format G723
Found description format PCMU
Found description format PCMA
Capabilities: us -
0x7fffffff(G723|GSM|ULAW|ALAW|G726|ADPCM|SLINR|LPC10|G729A|SPEEX|ILBC|UNKN|UNKN,
peer - audio=0x10d(G723|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined -
0x10d(G723|ULAW|ALAW|G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined -
0x0(EMPTY)Found peer '10916'
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
219.xx.xx.9:5060;branch=z9hG4bK8HxS8oAgT;received=219.xx.xx.9;rport=5060
From: "10916"
<sip:219.xx.xx.9@60.xx.xx.164>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:192.168.1.2@60.xx.xx.164>;tag=as56f2bbc0
Call-ID: BkmCUMXIAe0otg6w@219.xx.xx.9
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:192.168.1.2@60.xx.xx.164>
Proxy-Authenticate: Digest realm="asterisk",
nonce="1c7e2a70"
Content-Length: 0
to 219.xx.xx.9:5060
Scheduling destruction of call 'BkmCUMXIAe0otg6w@219.xx.xx.9' in 15000
ms
Sip read:
ACK sip:192.168.1.2@60.xx.xx.164 SIP/2.0
Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bK8HxS8oAgT
Max-Forwards: 70
User-Agent: PA168S
From: "10916"
<sip:219.xx.xx.9@60.xx.xx.164>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:192.168.1.2@60.xx.xx.164>;tag=as56f2bbc0
Call-ID: BkmCUMXIAe0otg6w@219.xx.xx.9
Contact: <sip:219.xx.xx.9@219.xx.xx.9:5060>
CSeq: 1 ACK
Content-Length: 0
10 headers, 0 lines
Sip read:
INVITE sip:192.168.1.2@60.xx.xx.164 SIP/2.0
Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bKt3iNpfZoe
Max-Forwards: 70
User-Agent: PA168S
From: "10916"
<sip:219.xx.xx.9@60.xx.xx.164>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:192.168.1.2@60.xx.xx.164>
Call-ID: BkmCUMXIAe0otg6w@219.xx.xx.9
Contact: <sip:219.xx.xx.9@219.xx.xx.9:5060>
Proxy-Authorization: Digest username="10916",
realm="asterisk",
nonce="1c7e2a70", uri="sip:10920@60.xx.xx.164",
response="9efb684d3aaffddf48b8857590c4b9b9"
CSeq: 2 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO,
PRACK, UPDATE
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 212
v=0
o=10916 91680803 00491630 IN IP4 219.xx.xx.9
s=SIP CALL
c=IN IP4 219.xx.xx.9
t=0 0
m=audio 5004 RTP/AVP 18 4 0 8
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
14 headers, 10 lines
Using latest request as basis request
Sending to 219.xx.xx.9 : 5060 (NAT)
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 219.xx.xx.9:5004
Found description format G729
Found description format G723
Found description format PCMU
Found description format PCMA
Capabilities: us -
0x7fffffff(G723|GSM|ULAW|ALAW|G726|ADPCM|SLINR|LPC10|G729A|SPEEX|ILBC|UNKN|UNKN,
peer - audio=0x10d(G723|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined -
0x10d(G723|ULAW|ALAW|G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined -
0x0(EMPTY)Found peer '10916'
Looking for 192.168.1.2 in default
Reliably Transmitting (NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
219.xx.xx.9:5060;branch=z9hG4bKt3iNpfZoe;received=219.xx.xx.9;rport=5060
From: "10916"
<sip:219.xx.xx.9@60.xx.xx.164>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:192.168.1.2@60.xx.xx.164>;tag=as56f2bbc0
Call-ID: BkmCUMXIAe0otg6w@219.xx.xx.9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:192.168.1.2@60.xx.xx.164>
Content-Length: 0
_________________________________________________________________
Express yourself instantly with MSN Messenger! Download today - it's FREE!
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
