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Shaun

2006-Apr-02 02:40 UTC

[Asterisk-Users] Cisco 7960 nat problems.

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@204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002"
<sip:1002@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002@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@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@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0@204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk",
nonce="74f7630a"
Content-Length: 0


---
Scheduling destruction of call
'00115cd9-d0370002-799a069f-51955597@192.168.1.102' in 15000 ms
Found user '1002'
localhost*CLI>
<-- SIP read from 68.5.xxx.xxx:51065:
INVITE sip:0@204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002"
<sip:1002@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002@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@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0@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@204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002"
<sip:1002@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002@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@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0@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@204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f
From: "1002"
<sip:1002@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:1002@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@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0@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@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0@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@204.10.xxx.xxx>;tag=00115cd9d0370002128504be-71e03bcb
To: <sip:0@204.10.xxx.xxx>;tag=as0ed772bf
Call-ID: 00115cd9-d0370002-799a069f-51955597@192.168.1.102
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0@204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk",
nonce="74f7630a"
Content-Length: 0


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


-- 

~Shaun
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Doug Lytle

2006-Apr-02 06:08 UTC

head link

[Asterisk-Users] Cisco 7960 nat problems.

Shaun wrote:> 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?You need to turn on the NAT support for the 7960 on the phone itself.

Settings --> SIP Configuration -->NAT Enabled (Y/N)

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."

Rich Adamson

2006-Apr-02 06:40 UTC

head link

[Asterisk-Users] Cisco 7960 nat problems.

Shaun wrote:> 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?
>  
I have several of these working just fine. You really do need nat=yes in 
sip.conf, and, change the 7960 config to say nat=yes in the phone.

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