[asterisk-users] Asterisk to Broadvoice SIP peer fails in 1.6.9-beta9

[David Siegel]

In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice.  Now, with the same SIP configuration, I cannot establish the peer.  I've enclosed a SIP log in the hope that someone can help me analyze this failure.  I'd guess the issue is NAT related and wondering if someone can spot a problem in the logs, below.

Some details to help read this log (I've changed these numbers for privacy purposes):

. My Asterisk server is behind a firewall.  It's internal address is 192.168.71.1.
. My public IP address is 123.123.123.123
. I am calling 2125551212
. My Broadvoice phone number is 9145551212

Here is the log:

  == Using SIP RTP CoS mark 5
    -- Executing [912125551234 at dialplan-siegel:1] Macro("SIP/siegeld-00e08e00", "dial-sip,12125551234 at sip.\
broadvoice.com") in new stack
    -- Executing [s at macro-dial-sip:1] Dial("SIP/siegeld-00e08e00", "SIP/12125551234 at sip.broadvoice.com") i\
n new stack
  == Using SIP RTP CoS mark 5
Audio is at 192.168.71.7 port 18596
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (NAT) to 123.123.123.123:5060:
INVITE sip:12125551234 at sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport
Max-Forwards: 70
From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4
To: <sip:12125551234 at sip.broadvoice.com>
Contact: <sip:9145551234 at 192.168.71.7>
Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Date: Mon, 30 Jun 2008 05:13:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 593017814 593017814 IN IP4 192.168.71.7
s=Asterisk PBX 1.6.0-beta9
c=IN IP4 192.168.71.7
t=0 0
m=audio 18596 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 12125551234 at sip.broadvoice.com
 stsca1*CLI>
<--- SIP read from UDP://123.123.123.123:5060 --->
SIP/2.0 100 Trying
Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com
CSeq: 102 INVITE
From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4
To: <sip:12125551234 at sip.broadvoice.com>
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
 stsca1*CLI>
<--- SIP read from UDP://123.123.123.123:5060 --->
SIP/2.0 403 Forbidden
Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com
CSeq: 102 INVITE
From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4
To: <sip:12125551234 at sip.broadvoice.com>;tag=lmno
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd
User-Agent: Asterisk PBX 1.6.0-beta9
Content-Type: application/sdp
Content-Length: 188
v=0
o=1213832004 593017814 593017814 IN IP4 192.168.71.7
s=-
c=IN IP4 192.168.71.7
t=0 0
m=audio 18596 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000

<------------->
--- (9 headers 9 lines) ---
Transmitting (NAT) to 123.123.123.123:5060:
ACK sip:12125551234 at sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport
Max-Forwards: 70
From: "David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4
To: <sip:12125551234 at sip.broadvoice.com>;tag=lmno
Contact: <sip:9145551234 at 192.168.71.7>
Call-ID: 6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta9
Content-Length: 0


---
[Jun 30 01:13:51] WARNING[3023]: chan_sip.c:14738 handle_response_invite: Received response: "Forbidden" f\
rom '"David Siegel" <sip:9145551234 at sip.broadvoice.com>;tag=as57923ac4'
    -- SIP/sip.broadvoice.com-00e0ddb0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s at macro-dial-sip:2] Goto("SIP/siegeld-00e08e00", "s-CONGESTION,1") in new stack
    -- Goto (macro-dial-sip,s-CONGESTION,1)
    -- Executing [s-CONGESTION at macro-dial-sip:1] PlayTones("SIP/siegeld-00e08e00", "congestion") in new st\
ack
    -- Auto fallthrough, channel 'SIP/siegeld-00e08e00' status is 'CONGESTION'
Really destroying SIP dialog '6f96b12763bae2bd7df963f02dbd2128 at sip.broadvoice.com' Method: INVI
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