Hello,
the unsucessfull test was made with kphone 3.12 and the sucessfull one
with 3.14. Try to upgrade the older version to 3.14 as well. Also try to
configure it the same way (i.e. use IP instead of 'asterix' in the
configuration).
Jan.
On 02-12 09:05, Jorge Merlino wrote:
Nils:
I'm afraid there is no more interesting dump to send you, kphone keeps sending
the same register message and SER keeps answering the same.
By the way yesterday I tried to register kphone without authentication
configured in SER and it didn't work either. Kphone keeps trying again and
again although it reiceives an OK answer from SER every time.
Oddly I've tested kphone with other proxies and always worked fine.
Here is a dump of this other situation (by the way I'm using kphone v. 3.12):
SipClient: Listening UDP on port: 5060
SipClient: Our address: 192.168.2.175
SipRegister: Auth is '(null)'
SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 09:00:38.025
--------------------------------
REGISTER sip:asterix SIP/2.0
Via: SIP/2.0/UDP 192.168.2.175
CSeq: 4764 REGISTER
To: "jorge" <sip:jorge@asterix>
Expires: 900
From: "jorge" <sip:jorge@asterix>
Call-ID: 123630670(a)192.168.2.175
Content-Length: 0
User-Agent: KPhone/3.12
Event: registration
Allow-Events: presence
Contact: "jorge"
<sip:jorge@192.168.2.175;transport=udp>;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.176:5060'
SipClient: Receiving message...
SipClient: Received: 09:00:38.041
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.175
CSeq: 4764 REGISTER
To: "jorge"
<sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22
From: "jorge" <sip:jorge@asterix>
Call-ID: 123630670(a)192.168.2.175
Contact: <sip:jorge@192.168.2.175:5062;transport=udp>;q=0.00;expires=337
Contact: <sip:jorge@192.168.2.175;transport=udp>;q=0.00;expires=900
Server: Sip EXpress router (0.8.11 (i386/linux))
Content-Length: 0
SipTransaction: Retransmit 1 (4000)
SipClient: Sending: 09:00:42.026
--------------------------------
REGISTER sip:asterix SIP/2.0
Via: SIP/2.0/UDP 192.168.2.175
CSeq: 4764 REGISTER
To: "jorge" <sip:jorge@asterix>
Expires: 900
From: "jorge" <sip:jorge@asterix>
Call-ID: 123630670(a)192.168.2.175
Content-Length: 0
User-Agent: KPhone/3.12
Event: registration
Allow-Events: presence
Contact: "jorge"
<sip:jorge@192.168.2.175;transport=udp>;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 09:00:42.038
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.175
CSeq: 4764 REGISTER
To: "jorge"
<sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22
From: "jorge" <sip:jorge@asterix>
Call-ID: 123630670(a)192.168.2.175
Contact: <sip:jorge@192.168.2.175:5062;transport=udp>;q=0.00;expires=333
Contact: <sip:jorge@192.168.2.175;transport=udp>;q=0.00;expires=900
Server: Sip EXpress router (0.8.11 (i386/linux))
Content-Length: 0
-----------------------------------------------------------------------------------------------------
Here is an example of a successful connection with other proxy (asterisk in
this case)
SipClient: Listening UDP on port: 5062
SipClient: Our address: 192.168.2.176
SipRegister: Auth is '(null)'
SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 08:47:08.026
--------------------------------
REGISTER sip:192.168.2.175 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.176:5062;rport
CSeq: 1124 REGISTER
To: "Luis Vazquez" <sip:101@192.168.2.175>
Expires: 600
From: "Luis Vazquez" <sip:101@192.168.2.175>
Call-ID: 546198833(a)192.168.2.176
Content-Length: 0
User-Agent: KPhone/3.14
Event: registration
Allow-Events: presence
Contact: "Luis Vazquez"
<sip:luis@192.168.2.176:5062;transport=udp>;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.175:5060'
SipClient: Receiving message...
SipClient: Received: 08:47:08.048
---------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.176:5062;rport
From: "Luis Vazquez" <sip:101@192.168.2.175>
To: "Luis Vazquez" <sip:101@192.168.2.175>;tag=as3f7be343
Call-ID: 546198833(a)192.168.2.176
CSeq: 1124 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101@192.168.2.175>
Content-Length: 0
SipCall: Incoming response
SipTransaction: Incoming Response
SipClient: Receiving message...
SipClient: Received: 08:47:08.051
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.176:5062;rport
From: "Luis Vazquez" <sip:101@192.168.2.175>
To: "Luis Vazquez" <sip:101@192.168.2.175>;tag=as3f7be343
Call-ID: 546198833(a)192.168.2.176
CSeq: 1124 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 600
Contact: <sip:101@192.168.2.175>;expires=600
Date: Tue, 02 Dec 2003 11:46:29 GMT
Content-Length: 0
SipCall: Incoming response
SipCall: Checking for Contact and Record-Route
SipCall: Setting Contact for this Call Member
SipTransaction: Incoming Response
ReRegistrationTimer (ms): 540000
Regards
Jorge
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