[asterisk-users] Problem with SIP phone outside local network

[Carlos Chavez]

	I am having a strange problem with an external SIP phone.  It can
register and receive calls but it cannot initiate any calls.  A
softphone on the same network works without problems.

	As far as I can notice the difference is that the hard phone is not
sending the proper contact info.  In the fullcontact field I can see its
private IP address "sip:1008 at 192.168.2.18:5060^3Btransport=udp" while
the softphone provides the public IP.  The hard phone is an Aastra
6730i.  A similar phone can make and receive calls when connected from
another external network so I do not think it is an Aastra issue.
Asterisk is behind a NAT (on DMZ) and has the proper externhost.  The
sip phone definition has nat=yes

	Any ideas?

	Here is a sip debug of the failed call:
<--- SIP read from UDP:201.141.67.189:41528 --->
INVITE sip:1007 at pbxwbu.xxxxx.org:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6
Max-Forwards: 70
From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f
To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>
Call-ID: 9567bd1f0b345f53
CSeq: 29187 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "1008" <sip:1008 at 192.168.2.18:5060;transport=udp>;
+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D21B027>"
Supported: path, 100rel, replaces
User-Agent: Aastra 6730i/3.2.2.1136
Content-Type: application/sdp
Content-Length: 595

v=0
o=MxSIP 0 1 IN IP4 192.168.2.18
s=SIP Call
c=IN IP4 192.168.2.18
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 25 lines) ---
Sending to 201.141.67.189:41528 (NAT)
Using INVITE request as basis request - 9567bd1f0b345f53
Found peer '1008' for '1008' from 201.141.67.189:41528

<--- Reliably Transmitting (NAT) to 201.141.67.189:41528 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.18;branch=z9hG4bKa522fd15e489606f6;received=201.141.67.189;rport=41528
From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f
To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46
Call-ID: 9567bd1f0b345f53
CSeq: 29187 INVITE
Server: Asterisk PBX 1.8.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="553cdc38"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '9567bd1f0b345f53' in 7040 ms
(Method: INVITE)
Retransmitting #1 (NAT) to 201.141.67.189:41528:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.18;branch=z9hG4bKa522fd15e489606f6;received=201.141.67.189;rport=41528
From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f
To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46
Call-ID: 9567bd1f0b345f53
CSeq: 29187 INVITE
Server: Asterisk PBX 1.8.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="553cdc38"
Content-Length: 0


---

<--- SIP read from UDP:201.141.67.189:41528 --->
ACK sip:1007 at pbxwbu.xxxxx.org:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6
Max-Forwards: 70
From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f
To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46
Call-ID: 9567bd1f0b345f53
CSeq: 29187 ACK
User-Agent: Aastra 6730i/3.2.2.1136
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:201.141.67.189:41528 --->
ACK sip:1007 at pbxwbu.xxxxx.org:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6
Max-Forwards: 70
From: "1008" <sip:1008 at pbxwbu.xxxxx.org:5060>;tag=fc268bfd9f
To: <sip:1007 at pbxwbu.xxxxx.org:5060;user=phone>;tag=as383d0a46
Call-ID: 9567bd1f0b345f53
CSeq: 29187 ACK
User-Agent: Aastra 6730i/3.2.2.1136
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
  == Using SIP RTP CoS mark 5


-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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