<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Arial">I've got an issue where I am trying to route calls
between Asterisk Servers. I can route calls inbound to a server but
seem to have an authentication issue going out over the same sip
account. It appears that my server isn't sending the second invite
after proxy authentication request. I can't figure out why; any ideas
would be greatly appreciated. Thanks!<br>
<br>
- Josh<br>
<br>
<br>
Here is my sip.conf:<br>
<br>
[general]<br>
context = default<br>
allowoverlap = no<br>
bindport = 5060<br>
bindaddr = 0.0.0.0<br>
srvlookup = yes<br>
externip = 172.21.235.2<br>
localnet = 172.21.235.2/255.255.0.0<br>
dtmfmode = rfc2833<br>
relaxdtmf = yes<br>
disallow = all<br>
allow = ulaw<br>
allow = gsm<br>
maxexpirey = 30<br>
defaultexpirey = 180<br>
relaxdtmf=yes<br>
canreinvite = no<br>
nat = 0<br>
UserAgent = Asterisk<br>
echocancel = yes<br>
echocancelwhenbridge = yes<br>
t38pt_udptl = no<br>
<br>
[trunk]<br>
type = friend<br>
callwaiting = yes<br>
caller id =<br>
contact =<br>
context = default<br>
fullname =<br>
group =<br>
hasagent = no<br>
hasdirectory = yes<br>
hasiax = no<br>
hasmanager = no<br>
hassip = yes<br>
host = 172.21.235.1<br>
secret = [password]<br>
threewaycalling = yes<br>
registersip = yes<br>
canreinvite = no<br>
nat = no<br>
dtmfmode = rfc2833<br>
registeriax = no<br>
disallow = all<br>
allow = gsm<br>
register=>trunk:[<a class="moz-txt-link-abbreviated" href="mailto:password]@172.21.235.1">password]@172.21.235.1</a><br>
<br>
<br>
Here is the applicable portion of extensions.conf:<br>
<br>
[default]<br>
exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)<br>
<br>
<br>
Here is the SIP Debug output:<br>
<br>
INVITE <a class="moz-txt-link-freetext" href="sip:510@172.21.235.1">sip:510@172.21.235.1</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport<br>
From: "Marci" <a class="moz-txt-link-rfc2396E" href="sip:3874@172.21.235.2"><sip:3874@172.21.235.2></a>;tag=as5951033c<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:510@172.21.235.1"><sip:510@172.21.235.1></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:3874@172.21.235.2"><sip:3874@172.21.235.2></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:430c49156ce4a7500b1fa57807b5acf1@172.21.235.2">430c49156ce4a7500b1fa57807b5acf1@172.21.235.2</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk<br>
Max-Forwards: 70<br>
Date: Tue, 30 Jun 2009 19:09:25 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
Content-Type: application/sdp<br>
Content-Length: 239<br>
<br>
v=0<br>
o=root 11411 11411 IN IP4 172.21.235.2<br>
s=session<br>
c=IN IP4 172.21.235.2<br>
t=0 0<br>
m=audio 11486 RTP/AVP 3 101<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
---<br>
^@<br>
^[[KWBPBXFG000304*CLI><br>
<--- SIP read from 172.21.235.1:5060 ---><br>
SIP/2.0 407 Proxy Authentication Required<br>
Via: SIP/2.0/UDP
172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060<br>
From: "Marci" <a class="moz-txt-link-rfc2396E" href="sip:3874@172.21.235.2"><sip:3874@172.21.235.2></a>;tag=as5951033c<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:510@172.21.235.1"><sip:510@172.21.235.1></a>;tag=as045cd609<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:430c49156ce4a7500b1fa57807b5acf1@172.21.235.2">430c49156ce4a7500b1fa57807b5acf1@172.21.235.2</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="4c4374da"<br>
Content-Length: 0<br>
<br>
<br>
<-------------><br>
^@<br>
^[[KWBPBXFG000304*CLI><br>
--- (11 headers 0 lines) ---<br>
^@<br>
^[[KWBPBXFG000304*CLI><br>
Transmitting (NAT) to 172.21.235.1:5060:<br>
ACK <a class="moz-txt-link-freetext" href="sip:510@172.21.235.1">sip:510@172.21.235.1</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport<br>
From: "Marci" <a class="moz-txt-link-rfc2396E" href="sip:3874@172.21.235.2"><sip:3874@172.21.235.2></a>;tag=as5951033c<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:510@172.21.235.1"><sip:510@172.21.235.1></a>;tag=as045cd609<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:3874@172.21.235.2"><sip:3874@172.21.235.2></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:430c49156ce4a7500b1fa57807b5acf1@172.21.235.2">430c49156ce4a7500b1fa57807b5acf1@172.21.235.2</a><br>
CSeq: 102 ACK<br>
User-Agent: Asterisk<br>
Max-Forwards: 70<br>
Content-Length: 0<br>
<br>
---<br>
^@<br>
^[[KWBPBXFG000304*CLI><br>
[Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253
handle_response_invite: ^@Failed to authenticate on INVITE to '"Marci"
<a class="moz-txt-link-rfc2396E" href="sip:3874@172.21.235.2"><sip:3874@172.21.235.2></a>;tag=as5951033c'<br>
^@<br>
^[[KWBPBXFG000304*CLI><br>
Really destroying SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:430c49156ce4a7500b1fa57807b5acf1@172.21.235.2">430c49156ce4a7500b1fa57807b5acf1@172.21.235.2</a>' Method: INVITE<br>
^@<br>
^[[KWBPBXFG000304*CLI><br>
Really destroying SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:0fe5f50f7674160d2ab3522f09060d46@127.0.0.1">0fe5f50f7674160d2ab3522f09060d46@127.0.0.1</a>' Method: REGISTER<br>
^@<br>
^[[KWBPBXFG000304*CLI><br>
<br>
</font>
</body>
</html>