<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=Content-Type content="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 12 (filtered medium)">
<style>
<!--
/* Font Definitions */
@font-face
{font-family:Wingdings;
panose-1:5 0 0 0 0 0 0 0 0 0;}
@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:11.0pt;
font-family:"Calibri","sans-serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
p.MsoListParagraph, li.MsoListParagraph, div.MsoListParagraph
{mso-style-priority:34;
margin-top:0in;
margin-right:0in;
margin-bottom:0in;
margin-left:.5in;
margin-bottom:.0001pt;
font-size:11.0pt;
font-family:"Calibri","sans-serif";}
span.EmailStyle17
{mso-style-type:personal-compose;
font-family:"Calibri","sans-serif";
color:windowtext;}
.MsoChpDefault
{mso-style-type:export-only;}
@page Section1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.Section1
{page:Section1;}
/* List Definitions */
@list l0
{mso-list-id:847409920;
mso-list-type:hybrid;
mso-list-template-ids:620817126 67698689 67698691 67698693 67698689 67698691 67698693 67698689 67698691 67698693;}
@list l0:level1
{mso-level-number-format:bullet;
mso-level-text:\F0B7;
mso-level-tab-stop:none;
mso-level-number-position:left;
text-indent:-.25in;
font-family:Symbol;}
@list l1
{mso-list-id:1680155374;
mso-list-type:hybrid;
mso-list-template-ids:1774606336 -228450334 67698691 67698693 67698689 67698691 67698693 67698689 67698691 67698693;}
@list l1:level1
{mso-level-start-at:0;
mso-level-number-format:bullet;
mso-level-text:-;
mso-level-tab-stop:none;
mso-level-number-position:left;
text-indent:-.25in;
font-family:"Calibri","sans-serif";
mso-fareast-font-family:Calibri;
mso-bidi-font-family:"Times New Roman";}
ol
{margin-bottom:0in;}
ul
{margin-bottom:0in;}
-->
</style>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=EN-US link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal>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.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Some details to help read this log (I’ve changed these
numbers for privacy purposes):<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>. My Asterisk server is behind a firewall. It’s
internal address is 192.168.71.1.<o:p></o:p></p>
<p class=MsoNormal>. My public IP address is 123.123.123.123<o:p></o:p></p>
<p class=MsoNormal>. I am calling 2125551212<o:p></o:p></p>
<p class=MsoNormal>. My Broadvoice phone number is 9145551212<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Here is the log:<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal> == Using SIP RTP CoS mark 5<o:p></o:p></p>
<p class=MsoNormal> -- Executing
[912125551234@dialplan-siegel:1] Macro("SIP/siegeld-00e08e00",
"dial-sip,12125551234@sip.\<o:p></o:p></p>
<p class=MsoNormal>broadvoice.com") in new stack<o:p></o:p></p>
<p class=MsoNormal> -- Executing [s@macro-dial-sip:1]
Dial("SIP/siegeld-00e08e00",
"SIP/12125551234@sip.broadvoice.com") i\<o:p></o:p></p>
<p class=MsoNormal>n new stack<o:p></o:p></p>
<p class=MsoNormal> == Using SIP RTP CoS mark 5<o:p></o:p></p>
<p class=MsoNormal>Audio is at 192.168.71.7 port 18596<o:p></o:p></p>
<p class=MsoNormal>Adding codec 0x4 (ulaw) to SDP<o:p></o:p></p>
<p class=MsoNormal>Adding codec 0x2 (gsm) to SDP<o:p></o:p></p>
<p class=MsoNormal>Adding codec 0x8 (alaw) to SDP<o:p></o:p></p>
<p class=MsoNormal>Reliably Transmitting (NAT) to 123.123.123.123:5060:<o:p></o:p></p>
<p class=MsoNormal>INVITE sip:12125551234@sip.broadvoice.com SIP/2.0<o:p></o:p></p>
<p class=MsoNormal>Via: SIP/2.0/UDP
192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport<o:p></o:p></p>
<p class=MsoNormal>Max-Forwards: 70<o:p></o:p></p>
<p class=MsoNormal>From: "David Siegel"
<sip:9145551234@sip.broadvoice.com>;tag=as57923ac4<o:p></o:p></p>
<p class=MsoNormal>To: <sip:12125551234@sip.broadvoice.com><o:p></o:p></p>
<p class=MsoNormal>Contact: <sip:9145551234@192.168.71.7><o:p></o:p></p>
<p class=MsoNormal>Call-ID: 6f96b12763bae2bd7df963f02dbd2128@sip.broadvoice.com<o:p></o:p></p>
<p class=MsoNormal>CSeq: 102 INVITE<o:p></o:p></p>
<p class=MsoNormal>User-Agent: Asterisk PBX 1.6.0-beta9<o:p></o:p></p>
<p class=MsoNormal>Date: Mon, 30 Jun 2008 05:13:51 GMT<o:p></o:p></p>
<p class=MsoNormal>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY<o:p></o:p></p>
<p class=MsoNormal>Supported: replaces, timer<o:p></o:p></p>
<p class=MsoNormal>Content-Type: application/sdp<o:p></o:p></p>
<p class=MsoNormal>Content-Length: 256<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>v=0<o:p></o:p></p>
<p class=MsoNormal>o=root 593017814 593017814 IN IP4 192.168.71.7<o:p></o:p></p>
<p class=MsoNormal>s=Asterisk PBX 1.6.0-beta9<o:p></o:p></p>
<p class=MsoNormal>c=IN IP4 192.168.71.7<o:p></o:p></p>
<p class=MsoNormal>t=0 0<o:p></o:p></p>
<p class=MsoNormal>m=audio 18596 RTP/AVP 0 3 8<o:p></o:p></p>
<p class=MsoNormal>a=rtpmap:0 PCMU/8000<o:p></o:p></p>
<p class=MsoNormal>a=rtpmap:3 GSM/8000<o:p></o:p></p>
<p class=MsoNormal>a=rtpmap:8 PCMA/8000<o:p></o:p></p>
<p class=MsoNormal>a=silenceSupp:off - - - -<o:p></o:p></p>
<p class=MsoNormal>a=ptime:20<o:p></o:p></p>
<p class=MsoNormal>a=sendrecv<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>---<o:p></o:p></p>
<p class=MsoNormal> -- Called 12125551234@sip.broadvoice.com<o:p></o:p></p>
<p class=MsoNormal> stsca1*CLI><o:p></o:p></p>
<p class=MsoNormal><--- SIP read from UDP://123.123.123.123:5060 ---><o:p></o:p></p>
<p class=MsoNormal>SIP/2.0 100 Trying<o:p></o:p></p>
<p class=MsoNormal>Call-ID: 6f96b12763bae2bd7df963f02dbd2128@sip.broadvoice.com<o:p></o:p></p>
<p class=MsoNormal>CSeq: 102 INVITE<o:p></o:p></p>
<p class=MsoNormal>From: "David Siegel"
<sip:9145551234@sip.broadvoice.com>;tag=as57923ac4<o:p></o:p></p>
<p class=MsoNormal>To: <sip:12125551234@sip.broadvoice.com><o:p></o:p></p>
<p class=MsoNormal>Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd<o:p></o:p></p>
<p class=MsoNormal>Content-Length: 0<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><-------------><o:p></o:p></p>
<p class=MsoNormal>--- (7 headers 0 lines) ---<o:p></o:p></p>
<p class=MsoNormal> stsca1*CLI><o:p></o:p></p>
<p class=MsoNormal><--- SIP read from UDP://123.123.123.123:5060 ---><o:p></o:p></p>
<p class=MsoNormal>SIP/2.0 403 Forbidden<o:p></o:p></p>
<p class=MsoNormal>Call-ID: 6f96b12763bae2bd7df963f02dbd2128@sip.broadvoice.com<o:p></o:p></p>
<p class=MsoNormal>CSeq: 102 INVITE<o:p></o:p></p>
<p class=MsoNormal>From: "David Siegel"
<sip:9145551234@sip.broadvoice.com>;tag=as57923ac4<o:p></o:p></p>
<p class=MsoNormal>To: <sip:12125551234@sip.broadvoice.com>;tag=lmno<o:p></o:p></p>
<p class=MsoNormal>Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd<o:p></o:p></p>
<p class=MsoNormal>User-Agent: Asterisk PBX 1.6.0-beta9<o:p></o:p></p>
<p class=MsoNormal>Content-Type: application/sdp<o:p></o:p></p>
<p class=MsoNormal>Content-Length: 188<o:p></o:p></p>
<p class=MsoNormal>v=0<o:p></o:p></p>
<p class=MsoNormal>o=1213832004 593017814 593017814 IN IP4 192.168.71.7<o:p></o:p></p>
<p class=MsoNormal>s=-<o:p></o:p></p>
<p class=MsoNormal>c=IN IP4 192.168.71.7<o:p></o:p></p>
<p class=MsoNormal>t=0 0<o:p></o:p></p>
<p class=MsoNormal>m=audio 18596 RTP/AVP 0 3 8<o:p></o:p></p>
<p class=MsoNormal>a=rtpmap:0 PCMU/8000<o:p></o:p></p>
<p class=MsoNormal>a=rtpmap:3 GSM/8000<o:p></o:p></p>
<p class=MsoNormal>a=rtpmap:8 PCMA/8000<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><-------------><o:p></o:p></p>
<p class=MsoNormal>--- (9 headers 9 lines) ---<o:p></o:p></p>
<p class=MsoNormal>Transmitting (NAT) to 123.123.123.123:5060:<o:p></o:p></p>
<p class=MsoNormal>ACK sip:12125551234@sip.broadvoice.com SIP/2.0<o:p></o:p></p>
<p class=MsoNormal>Via: SIP/2.0/UDP
192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport<o:p></o:p></p>
<p class=MsoNormal>Max-Forwards: 70<o:p></o:p></p>
<p class=MsoNormal>From: "David Siegel"
<sip:9145551234@sip.broadvoice.com>;tag=as57923ac4<o:p></o:p></p>
<p class=MsoNormal>To: <sip:12125551234@sip.broadvoice.com>;tag=lmno<o:p></o:p></p>
<p class=MsoNormal>Contact: <sip:9145551234@192.168.71.7><o:p></o:p></p>
<p class=MsoNormal>Call-ID: 6f96b12763bae2bd7df963f02dbd2128@sip.broadvoice.com<o:p></o:p></p>
<p class=MsoNormal>CSeq: 102 ACK<o:p></o:p></p>
<p class=MsoNormal>User-Agent: Asterisk PBX 1.6.0-beta9<o:p></o:p></p>
<p class=MsoNormal>Content-Length: 0<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>---<o:p></o:p></p>
<p class=MsoNormal>[Jun 30 01:13:51] WARNING[3023]: chan_sip.c:14738
handle_response_invite: Received response: "Forbidden" f\<o:p></o:p></p>
<p class=MsoNormal>rom '"David Siegel"
<sip:9145551234@sip.broadvoice.com>;tag=as57923ac4'<o:p></o:p></p>
<p class=MsoNormal> -- SIP/sip.broadvoice.com-00e0ddb0 is
circuit-busy<o:p></o:p></p>
<p class=MsoNormal> == Everyone is busy/congested at this time (1:0/1/0)<o:p></o:p></p>
<p class=MsoNormal> -- Executing [s@macro-dial-sip:2]
Goto("SIP/siegeld-00e08e00", "s-CONGESTION,1") in new stack<o:p></o:p></p>
<p class=MsoNormal> -- Goto (macro-dial-sip,s-CONGESTION,1)<o:p></o:p></p>
<p class=MsoNormal> -- Executing
[s-CONGESTION@macro-dial-sip:1] PlayTones("SIP/siegeld-00e08e00",
"congestion") in new st\<o:p></o:p></p>
<p class=MsoNormal>ack<o:p></o:p></p>
<p class=MsoNormal> -- Auto fallthrough, channel
'SIP/siegeld-00e08e00' status is 'CONGESTION'<o:p></o:p></p>
<p class=MsoNormal>Really destroying SIP dialog
'6f96b12763bae2bd7df963f02dbd2128@sip.broadvoice.com' Method: INVI<o:p></o:p></p>
</div>
</body>
</html>