[Asterisk-Users] Asterisk not paying attention to NAT Setting

[Matthew Boehm]

Perhaps someone else can make sense of this but I can't figure out why
asterisk is not paying attention to the NAT setting. Asterisk keeps trying
to use the UA's internal address to respond too.

We have 10 of these phones in another situation and they work fine.

The originating IP address (the UA) is 22.22.22.22 and the asterisk server
is 33.33.33.33. The UA is behind a NAT (Linksys BEFSR41 router, using the
192.168.1.0 network)

asterisk*CLI> sip show peer sales

  * Name       : sales
  Nat          : Always
  Addr->IP     : 22.22.22.22 Port 15060
  Defaddr->IP  : 0.0.0.0 Port 8302
  Def. Username: sales
  Codecs       : 0x10c (ulaw|alaw|g729)
  Codec Order  : (g729|ulaw|alaw)
  Status       : UNKNOWN
  Useragent    : IPCall104 02.09.20
  Reg. Contact : sip:sales at 192.168.1.101:5060


U 22.22.22.22:15060 -> 33.33.33.33:5060
INVITE sip:3044 at 33.33.33.33:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK_00D0E90184A4_T685AB1DB.
Session-Expires: 1800.
From: "sales" <sip:sales at 33.33.33.33:5060>;tag=00D0E90184A4_T1113797725.
To: <sip:3044 at 33.33.33.33:5060>.
Call-ID: CALL_ID7_00D0E90184A4_T489152993 at 192.168.1.101.
CSeq: 1211705509 INVITE.
Contact: <sip:sales at 192.168.1.101:5060>.
Max-Forwards: 70.
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO.
Supported: 100rel,timer,replaces.
User-Agent: IPCall104 02.09.20.
Content-Type: application/sdp.
Content-Length: 244.
.
v=0.
o=sales 271633486 271633486 IN IP4 192.168.1.101.
s=IPCall104 02.09.20.
c=IN IP4 192.168.1.101.
t=0 0.
m=audio 41000 RTP/AVP 0 18 4.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000/1.
a=sendrecv.

U 33.33.33.33:5060 -> 192.168.1.101:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK_00D0E90184A4_T685AB1DB.
From: "sales" <sip:sales at 33.33.33.33:5060>;tag=00D0E90184A4_T1113797725.
To: <sip:3044 at 33.33.33.33:5060>;tag=as7fd8eff6.
Call-ID: CALL_ID7_00D0E90184A4_T489152993 at 192.168.1.101.
CSeq: 1211705509 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:3044 at 33.33.33.33>.
Proxy-Authenticate: Digest realm="asterisk", nonce="2d8c4966".
Content-Length: 0.
.

U 22.22.22.22:15060 -> 33.33.33.33:5060
INVITE sip:3044 at 33.33.33.33:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK_00D0E90184A4_T685AB1DB.
Session-Expires: 1800.
From: "sales" <sip:sales at 33.33.33.33:5060>;tag=00D0E90184A4_T1113797725.
To: <sip:3044 at 33.33.33.33:5060>.
Call-ID: CALL_ID7_00D0E90184A4_T489152993 at 192.168.1.101.
CSeq: 1211705509 INVITE.
Contact: <sip:sales at 192.168.1.101:5060>.
Max-Forwards: 70.
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO.
Supported: 100rel,timer,replaces.
User-Agent: IPCall104 02.09.20.
Content-Type: application/sdp.
Content-Length: 244.
.
v=0.
o=sales 271633486 271633486 IN IP4 192.168.1.101.
s=IPCall104 02.09.20.
c=IN IP4 192.168.1.101.
t=0 0.
m=audio 41000 RTP/AVP 0 18 4.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000/1.
a=sendrecv.

U 33.33.33.33:5060 -> 192.168.1.101:5060
SIP/2.0 488 Not Acceptable Here (codec error).
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK_00D0E90184A4_T685AB1DB.
From: "sales" <sip:sales at 33.33.33.33:5060>;tag=00D0E90184A4_T1113797725.
To: <sip:3044 at 33.33.33.33:5060>;tag=as7fd8eff6.
Call-ID: CALL_ID7_00D0E90184A4_T489152993 at 192.168.1.101.
CSeq: 1211705509 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:3044 at 33.33.33.33>.
Content-Length: 0.
.

U 33.33.33.33:5060 -> 192.168.1.101:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK_00D0E90184A4_T685AB1DB.
From: "sales" <sip:sales at 33.33.33.33:5060>;tag=00D0E90184A4_T1113797725.
To: <sip:3044 at 33.33.33.33:5060>;tag=as7fd8eff6.
Call-ID: CALL_ID7_00D0E90184A4_T489152993 at 192.168.1.101.
CSeq: 1211705509 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:3044 at 33.33.33.33>.
Proxy-Authenticate: Digest realm="asterisk", nonce="2d8c4966".
Content-Length: 0.
.


Thanks for help,
Matthew

-- 
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Matthew Boehm, IT Director                    Cypress Telecommunications
mboehm at cytelcom.com                   3838 N. Sam Houston Parkway E #400
T: 832-200-8640 x3044                                  Houston, TX 77032

My girlfriend was recently diagnosed with multiple personality disorder;
 When she called yesterday, my CallerID box exploded.
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