[asterisk-users] Problem connecting to another server, Failed to authenticate on INVITE

Sun Jun 15 22:18:10 CDT 2008

When I call from SJPhone (softphone) and connect to an asterisk server
(source.asterisk.server), then dial an extension, which connects to a
different asterisk server (destination.asterisk.server), it fails.

"chan_sip.c:12253 handle_response_invite: Failed to authenticate on INVITE
to 'source.asterisk.server.ip'".

SIP Debug shows that the destination server is asking for proxy
authentication.

I can connect from the soft phone to the source asterisk server and dial an
extension which runs an AGI application on that server without problems.

I can connect from the soft phone to the 111 at destination.asterisk.server
SIP address with no problem.

The source and destination asterisk servers are not using NAT. The soft
phone is behind NAT.

Configuration files and logs are below. Any ideas on how I can
successfully make this connection?

; source.asterisk.server, sip.conf:
[ccn]
srvlookup=yes
type=user
secret=<password>
qualify=yes
nat=yes
dtmfmode=rfc2833
host=dynamic
canreinvite=no
context=ccn_in
disallow=all
allow=ulaw

; source.asterisk.server, extensions.conf:
[ccn_in]
exten => 111,1,wait(1)
exten => 111,n,DISA(no-password,internal) 
[internal]
include => outbound
include => default 
[default]
exten => 112,1,dial(sip/111 at destination.asterisk.server)

; destination.asterisk.server, sip.conf:
[111]
disallow=all
allow=ulaw
type=peer 
dtmfmode=rfc2833 
context=ctx
insecure=port,invite
nat=no

; destination.asterisk.server, extensions.conf:
[ctx]
exten => 111,1,answer()
exten => 111,n,wait(1)
exten => 111,n,agi(script.agi)
exten => 111,n,hangup 

---Start Transcript---

<--- SIP read from client.external.ip:5060 --->
INVITE sip:111 at source.asterisk.server SIP/2.0
Via: SIP/2.0/UDP
client.internal.ip;rport;branch=z9hG4bKc0a8017b000000c74855cc160000292700000393
Content-Length: 339
Contact: <sip:ccn at client.internal.ip:5060>
Call-ID: D41F89F7-99AD-405D-9AC1-32412952751A at client.internal.ip
Content-Type: application/sdp
CSeq: 1 INVITE
From: "Chris N"<sip:ccn at source.asterisk.server>;tag=5250369537080
Max-Forwards: 70
To: <sip:111 at source.asterisk.server>
User-Agent: SJphone/1.60.289a (SJ Labs)

v=0
o=- 3422571157 3422571157 IN IP4 client.internal.ip
s=SJphone
c=IN IP4 client.internal.ip
t=0 0
a=direction:active
m=audio 49160 RTP/AVP 0 3 97 98 8 101
a=rtpmap:0 PCMU/8000
...snip...
a=fmtp:101 0-11,16

<------------->
--- (11 headers 15 lines) ---
Sending to client.external.ip : 5060 (NAT)
Using INVITE request as basis request -
D41F89F7-99AD-405D-9AC1-32412952751A at client.internal.ip
Found user 'ccn'
Found RTP audio format 0
...snip...
Found RTP audio format 101
Peer audio RTP is at port client.internal.ip:49160
Found audio description format PCMU for ID 0
...snip...
Found audio description format iLBC for ID 98
Got unsupported a:fmtp in SDP offer 
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer 
Capabilities: us - 0x4 (ulaw), peer - audio=0x40e
(gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port client.internal.ip:49160
Looking for 111 in ccn_in (domain source.asterisk.server)
list_route: hop: <sip:ccn at client.internal.ip:5060>

<--- Transmitting (NAT) to client.external.ip:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
client.internal.ip;branch=z9hG4bKc0a8017b000000c74855cc160000292700000393;
received=client.external.ip;rport=5060
From: "Chris N"<sip:ccn at source.asterisk.server>;tag=5250369537080
To: <sip:111 at source.asterisk.server>
Call-ID: D41F89F7-99AD-405D-9AC1-32412952751A at client.internal.ip
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:111 at source.asterisk.server.ip>
Content-Length: 0

<------------>
-- Executing [111 at ccn_in:1] Wait("SIP/ccn-081c9260", "1") in new stack
-- Executing [111 at ccn_in:2] DISA("SIP/ccn-081c9260",
"no-password|internal") in new stack
Audio is at source.asterisk.server.ip port 15184
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to client.external.ip:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
client.internal.ip;branch=z9hG4bKc0a8017b000000c74855cc160000292700000393;
received=client.external.ip;rport=5060
From: "Chris N"<sip:ccn at source.asterisk.server>;tag=5250369537080
To: <sip:111 at source.asterisk.server>;tag=as69b98e07
Call-ID: D41F89F7-99AD-405D-9AC1-32412952751A at client.internal.ip
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:111 at source.asterisk.server.ip>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 19975 19975 IN IP4 source.asterisk.server.ip
s=session
c=IN IP4 source.asterisk.server.ip
t=0 0
m=audio 15184 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
...snip...
a=sendrecv

<------------>
tz*CLI> 
<--- SIP read from client.external.ip:5060 --->
ACK sip:111 at source.asterisk.server.ip SIP/2.0
Via: SIP/2.0/UDP
client.internal.ip;rport;branch=z9hG4bKc0a8017b000000c74855cc17000055d900000396
Content-Length: 0
Call-ID: D41F89F7-99AD-405D-9AC1-32412952751A at client.internal.ip
CSeq: 1 ACK
From: "Chris N"<sip:ccn at source.asterisk.server>;tag=5250369537080
Max-Forwards: 70
To: <sip:111 at source.asterisk.server>;tag=as69b98e07
User-Agent: SJphone/1.60.289a (SJ Labs)

<------------->
--- (9 headers 0 lines) ---
-- Executing [112 at internal:1] Dial("SIP/ccn-081c9260",
"sip/111 at destination.asterisk.server") in new stack
Audio is at source.asterisk.server.ip port 18784
Adding codec 0x4 (ulaw) to SDP
...snip...
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to destination.asterisk.server.ip:5060:
INVITE sip:111 at destination.asterisk.server SIP/2.0
Via: SIP/2.0/UDP
source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861;rport
From: "Chris N" <sip:ccn at source.asterisk.server.ip>;tag=as2c01a79e
To: <sip:111 at destination.asterisk.server>
Contact: <sip:ccn at source.asterisk.server.ip>
Call-ID: 5e4039434ba50dda6265742d18a88db8 at source.asterisk.server.ip
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 16 Jun 2008 02:12:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 19975 19975 IN IP4 source.asterisk.server.ip
s=session
c=IN IP4 source.asterisk.server.ip
t=0 0
m=audio 18784 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
...snip...
a=sendrecv

---
-- Called 111 at destination.asterisk.server
tz*CLI> 
<--- SIP read from destination.asterisk.server.ip:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861;
received=source.asterisk.server.ip;rport=5060
From: "Chris N" <sip:ccn at source.asterisk.server.ip>;tag=as2c01a79e
To: <sip:111 at destination.asterisk.server>;tag=as6b3765e7
Call-ID: 5e4039434ba50dda6265742d18a88db8 at source.asterisk.server.ip
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="destination.asterisk.server", nonce="6cc9de0c"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to destination.asterisk.server.ip:5060:
ACK sip:111 at destination.asterisk.server SIP/2.0
Via: SIP/2.0/UDP
source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861;rport
From: "Chris N" <sip:ccn at source.asterisk.server.ip>;tag=as2c01a79e
To: <sip:111 at destination.asterisk.server>;tag=as6b3765e7
Contact: <sip:ccn at source.asterisk.server.ip>
Call-ID: 5e4039434ba50dda6265742d18a88db8 at source.asterisk.server.ip
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
[Jun 15 21:12:42] NOTICE[19999]: chan_sip.c:12253 handle_response_invite:
Failed to authenticate on INVITE to '"Chris N"
<sip:ccn at source.asterisk.server.ip>;tag=as2c01a79e'
-- SIP/destination.asterisk.server-081cfab0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/ccn-081c9260' status is 'CONGESTION'
---End Transcript---

-- 
Chris Nestrud
Email: ccn at panix.com
http://ChrisNestrud.com/