[Asterisk-Users] Strange sip address?

Dan Zhou Mon, 31 Jan 2005 13:34:58 -0800

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:[EMAIL PROTECTED]>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:[EMAIL PROTECTED]>

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:[EMAIL PROTECTED]>;tag=ks8tHkBudRSI7Ydz
To: "10916" <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 25463 REGISTER
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=ks8tHkBudRSI7Ydz To: "10916" <sip:[EMAIL PROTECTED]>;tag=as4016e46b Call-ID: [EMAIL PROTECTED] CSeq: 25463 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=ks8tHkBudRSI7Ydz To: "10916" <sip:[EMAIL PROTECTED]>;tag=as4016e46b Call-ID: [EMAIL PROTECTED] CSeq: 25463 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="44ae239c" Content-Length: 0


to 219.xx.xx.9:5060
Scheduling destruction of call '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED]>;tag=uL0iKjrTXppeLeoq To: "10916" <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 25464 REGISTER Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=uL0iKjrTXppeLeoq To: "10916" <sip:[EMAIL PROTECTED]>;tag=as4016e46b Call-ID: [EMAIL PROTECTED] CSeq: 25464 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=uL0iKjrTXppeLeoq To: "10916" <sip:[EMAIL PROTECTED]>;tag=as4016e46b Call-ID: [EMAIL PROTECTED] CSeq: 25464 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: <sip:[EMAIL PROTECTED]: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 '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED]>;tag=as4e42ecfb
To: <sip:219.xx.xx.9>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as4e42ecfb To: <sip:219.xx.xx.9>;tag=zIQDT3uvUSCwzF2y Contact: <sip:[EMAIL PROTECTED]: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 '[EMAIL PROTECTED]'

Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bK8HxS8oAgT Max-Forwards: 70 User-Agent: PA168S From: "10916" <sip:[EMAIL PROTECTED]>;tag=7Fqj0hOx08JI9aY3 To: "10920" <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=7Fqj0hOx08JI9aY3 To: "10920" <sip:[EMAIL PROTECTED]>;tag=as56f2bbc0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk", nonce="1c7e2a70" Content-Length: 0


to 219.xx.xx.9:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms


Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bK8HxS8oAgT
Max-Forwards: 70
User-Agent: PA168S
From: "10916" <sip:[EMAIL PROTECTED]>;tag=7Fqj0hOx08JI9aY3
To: "10920" <sip:[EMAIL PROTECTED]>;tag=as56f2bbc0
Call-ID: [EMAIL PROTECTED]
Contact: <sip:[EMAIL PROTECTED]:5060>
CSeq: 1 ACK
Content-Length: 0

10 headers, 0 lines

Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bKt3iNpfZoe Max-Forwards: 70 User-Agent: PA168S From: "10916" <sip:[EMAIL PROTECTED]>;tag=7Fqj0hOx08JI9aY3 To: "10920" <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]:5060> Proxy-Authorization: Digest username="10916", realm="asterisk", nonce="1c7e2a70", uri="sip:[EMAIL PROTECTED]", 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:[EMAIL PROTECTED]>;tag=7Fqj0hOx08JI9aY3 To: "10920" <sip:[EMAIL PROTECTED]>;tag=as56f2bbc0 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0

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