[asterisk-users] Authentication Issue Between Servers

[Joshua Billings]

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!

- Josh


Here is my sip.conf:

[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
externip = 172.21.235.2
localnet = 172.21.235.2/255.255.0.0
dtmfmode = rfc2833
relaxdtmf = yes
disallow = all
allow = ulaw
allow = gsm
maxexpirey = 30
defaultexpirey = 180
relaxdtmf=yes
canreinvite = no
nat = 0
UserAgent = Asterisk
echocancel = yes
echocancelwhenbridge = yes
t38pt_udptl = no

[trunk]
type = friend
callwaiting = yes
caller id =
contact =
context = default
fullname =
group =
hasagent = no
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
host = 172.21.235.1
secret = [password]
threewaycalling = yes
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
registeriax = no
disallow = all
allow = gsm
register=>trunk:[password]@172.21.235.1


Here is the applicable portion of extensions.conf:

[default]
exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)


Here is the SIP Debug output:

INVITE sip:510 at 172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
To: <sip:510 at 172.21.235.1>
Contact: <sip:3874 at 172.21.235.2>
Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Tue, 30 Jun 2009 19:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 11411 11411 IN IP4 172.21.235.2
s=session
c=IN IP4 172.21.235.2
t=0 0
m=audio 11486 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
^@
^[[KWBPBXFG000304*CLI>
<--- SIP read from 172.21.235.1:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060
From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
To: <sip:510 at 172.21.235.1>;tag=as045cd609
Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c4374da"
Content-Length: 0


<------------->
^@
^[[KWBPBXFG000304*CLI>
--- (11 headers 0 lines) ---
^@
^[[KWBPBXFG000304*CLI>
Transmitting (NAT) to 172.21.235.1:5060:
ACK sip:510 at 172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
To: <sip:510 at 172.21.235.1>;tag=as045cd609
Contact: <sip:3874 at 172.21.235.2>
Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0

---
^@
^[[KWBPBXFG000304*CLI>
[Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253 
handle_response_invite: ^@Failed to authenticate on INVITE to '"Marci" 
<sip:3874 at 172.21.235.2>;tag=as5951033c'
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog 
'430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2' Method: INVITE
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog 
'0fe5f50f7674160d2ab3522f09060d46 at 127.0.0.1' Method: REGISTER
^@
^[[KWBPBXFG000304*CLI>

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