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 [EMAIL PROTECTED]
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/[EMAIL PROTECTED])
; 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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
client.internal.ip;rport;branch=z9hG4bKc0a8017b000000c74855cc160000292700000393
Content-Length: 339
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
Content-Type: application/sdp
CSeq: 1 INVITE
From: "Chris N"<sip:[EMAIL PROTECTED]>;tag=5250369537080
Max-Forwards: 70
To: <sip:[EMAIL PROTECTED]>
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 -
[EMAIL PROTECTED]
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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=5250369537080
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
<------------>
-- Executing [EMAIL PROTECTED]:1] Wait("SIP/ccn-081c9260", "1") in new stack
-- Executing [EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=5250369537080
To: <sip:[EMAIL PROTECTED]>;tag=as69b98e07
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
client.internal.ip;rport;branch=z9hG4bKc0a8017b000000c74855cc17000055d900000396
Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
From: "Chris N"<sip:[EMAIL PROTECTED]>;tag=5250369537080
Max-Forwards: 70
To: <sip:[EMAIL PROTECTED]>;tag=as69b98e07
User-Agent: SJphone/1.60.289a (SJ Labs)
<------------->
--- (9 headers 0 lines) ---
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/ccn-081c9260",
"sip/[EMAIL PROTECTED]") 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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861;rport
From: "Chris N" <sip:[EMAIL PROTECTED]>;tag=as2c01a79e
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as2c01a79e
To: <sip:[EMAIL PROTECTED]>;tag=as6b3765e7
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861;rport
From: "Chris N" <sip:[EMAIL PROTECTED]>;tag=as2c01a79e
To: <sip:[EMAIL PROTECTED]>;tag=as6b3765e7
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;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: [EMAIL PROTECTED]
http://ChrisNestrud.com/
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