[Asterisk-Users] Cisco 7960 nat problems.

Sun Apr 2 02:40:48 MST 2006

I have a asterisk server running on site listening on a public ip.  Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed.  My home network is simple, Cox cable connection hooked to a linksys wrt router.  The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort.  I removed the linksys router and plugged the phone directly into the cable modem and now the phone can connect fine and works.  I pasted below the sip debug output, anybody know what's going on or have experience with this?

------------ sip.conf ----------------
[general]
context=default              
bindport=5060                 
bindaddr=0.0.0.0                
srvlookup=yes                  

[1002]
username=1002
secret=********
type=friend
host=dynamic
allow=all
context=default
nat=yes

---------- SIP DEBUG -------------
<-- SIP read from 68.5.xxx.xxx:51065:
INVITE sip:0 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002 at 192.168.1.102:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102
s=SIP Call
t=0 0
m=audio 25584 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (16 headers 13 lines)---
Using INVITE request as basis request - 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
Sending to 192.168.1.102 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 68.5.xxx.xxx:51065:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0 at 204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"
Content-Length: 0

---
Scheduling destruction of call '00115cd9-d0370002-799a069f-51955597 at 192.168.1.102' in 15000 ms
Found user '1002'
localhost*CLI>
<-- SIP read from 68.5.xxx.xxx:51065:
INVITE sip:0 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002 at 192.168.1.102:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102
s=SIP Call
t=0 0
m=audio 25584 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (16 headers 13 lines)---
Ignoring this INVITE request
Retransmitting #1 (NAT) to 68.5.xxx.xxx:51065:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0 at 204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"
Content-Length: 0

---
localhost*CLI>
<-- SIP read from 68.5.xxx.xxx:51065:
INVITE sip:0 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002 at 192.168.1.102:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102
s=SIP Call
t=0 0
m=audio 25584 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (16 headers 13 lines)---
Ignoring this INVITE request
Retransmitting #2 (NAT) to 68.5.xxx.xxx:51065:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0 at 204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"
Content-Length: 0

---
localhost*CLI>
<-- SIP read from 68.5.xxx.xxx:51065:
INVITE sip:0 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002 at 192.168.1.102:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102
s=SIP Call
t=0 0
m=audio 25584 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (16 headers 13 lines)---
Ignoring this INVITE request
Retransmitting #3 (NAT) to 68.5.xxx.xxx:51065:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0 at 204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"
Content-Length: 0

---
Retransmitting #4 (NAT) to 68.5.xxx.xxx:51065:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0 at 204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"
Content-Length: 0

---
Retransmitting #5 (NAT) to 68.5.xxx.xxx:51065:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx
From: "1002" <sip:1002 at 204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0 at 204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597 at 192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0 at 204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"
Content-Length: 0

---
Destroying call '00115cd9-d0370002-799a069f-51955597 at 192.168.1.102'

-- 

~Shaun
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060402/c4602319/attachment.htm