GXP-2000 Unregister Debug Help

#1
Hi all,

New to this forum but I have seen many posts from people relating to the GXP-2000 unregistering and possible suggesitons for solutions. So far, I have tried everything and I am running on the 1.1.1.14 of the firmware.

However, my problems persist. Therefore today I decided to go down the detective route and install a SYSLOG server, put the phone in debug mode and try and recreate the problem. I come from a developer background so bug analysis is a bit of a past time.

Basically, I rebooted the phone with the SYSLOG server running and debug switched on and I waited for the phone to unregister, then I trekked through the debug formatting the messages to try and find out what was going on. Now, I know nothing about SIP message formats or their content exactly, however some of the tracing information does make sense. The trace ran until such time as the phone unregistered. It is interesting as many of the posts suggest that a reboot solves the problem and indeed it does. The reason I believe this is the case is that the phone actually stops sending requests at a certain point and nothing seems to encourage it to retry again. Although I have not left this for a couple of hours as yet (in progress) so it may do and I will post the info here if it does.

This is probably the reason why some people think it is the phone and others the SIP provider.

Below, is a copy of the trace in its entirety, formatted for easier reading. Now I need some further help as I believe I have gone as far as I can. At the bottom of the trace information, the phone no longer appears to send requests until it is rebooted. So the last entry is the last time the phone attempted a request - there is nothing else in the log and the SYSLOG server just sits waiting for another request.

Code: Select all

syslog server(port:514) started on Sun Dec 03 16:23:41 2006 syslog 
server(port:514) started on Sun Dec 03 16:27:10 2006 GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Grandstream GXP2000 1.1.1.14 
1.1.1.5GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] Provision attempt 
1GS_LOG: [00:0b:82:08:9b:04][701][FF71][0101010E] File Not Found: 
/gs/cfg000b82089b04GS_LOG: [00:0b:82:08:9b:04][701][FF71][0101010E] File Not 
Found: /gs/boot55b.binGS_LOG: [00:0b:82:08:9b:04][701][FF71][0101010E] File Not 
Found: /gs/gxp2000b.binGS_LOG: [00:0b:82:08:9b:04][701][FF71][0101010E] File Not 
Found: /gs/ring1.binGS_LOG: [00:0b:82:08:9b:04][701][FF71][0101010E] File Not 
Found: /gs/ring2.binGS_LOG: [00:0b:82:08:9b:04][701][FF71][0101010E] File Not 
Found: /gs/ring3.binGS_LOG: 

[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER sip:sip.voipfone.co.uk 
SIP/2.0  Via: SIP/2.0/UDP 192.168.16.11:5060;branch=z9hG4bKcd897fb833a076ea 
From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Call-ID: 7ff47dc957ede49e@192.168.16.11 
CSeq: 10001 REGISTER  Expires: 3888000  User-Agent: Grandstream GXP2000 1.1.1.14 
Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 
195.189.173.12:5065 5060 543 SIP/2.0 401 Unauthorized  Via: SIP/2.0/UDP 
192.168.16.11:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bKcd897fb833a07
6ea  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>;tag=as192caff2  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10001 REGISTER  User-Agent: Voipfone Sip 
Network  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER  Contact: 
<sip>  WWW-Authenticate: Digest realm="asterisk", 
nonce="250bc905"  Content-Length: 0    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 401GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 11GS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER sip:sip.voipfone.co.uk 
SIP/2.0  Via: SIP/2.0/UDP 192.168.16.11:5060;branch=z9hG4bK34d65251d670225a
From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Authorization: Digest 
username="30125598*200", realm="asterisk", algorithm=MD5, 
uri="sip:sip.voipfone.co.uk", nonce="250bc905", 
response="5f1f160984712c227d093ffe9a80ae27"  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10002 REGISTER  Expires: 3888000  User-
Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 
195.189.173.12:5065 5060 653 SIP/2.0 200 OK  Via: SIP/2.0/UDP 
192.168.16.11:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK34d65251d6702
25a  Record-Route: <sip>  Record-Route: 
<sip>  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>;tag=as192caff2  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10002 REGISTER  User-Agent: Voipfone Sip 
Network  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER  Expires: 60  Contact: 
<sip>;expires=60  Date: Sun, 03 Dec 2006 
16:27:50 GMT  Content-Length: 0    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 200GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 11GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Alan Wyatt-Jones REGISTERED for 60 
seconds;re-REGISTER in 45 secondsGS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 8 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 
SUBSCRIBE sip:30125598*200@sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK1a39806f83f18055  From: "Alan Wyatt-Jones" 
<sip>;tag=952b1576fe27f353  To: 
<sip>  Contact: 
<sip>  Call-ID: d47d9907d9a619c9@192.168.16.11 
CSeq: 1001 SUBSCRIBE  User-Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70 
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Event: message-summary  Expires: 3888000  Accept: application/simple-message-
summary  Content-Length: 0          GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] 195.189.173.12:5065 5060 650 SIP/2.0 
501 Not Implemented  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK1a39806f83f18055;rport=5060;received=xx.xx.xx.xx  From: "Alan Wyatt-Jones" 
<sip>;tag=952b1576fe27f353  To: 
<sip>;tag=dae9f5855e38e874829059f7681d766b.2a11 
Call-ID: d47d9907d9a619c9@192.168.16.11  CSeq: 1001 SUBSCRIBE  Server: Sip 
EXpress router (0.8.99-dev12 (i386/linux))  Content-Length: 0  Warning: 392 
195.189.173.12:5065 "Noisy feedback tells:  pid=1405 req_src_ip=xx.xx.xx.xx 
req_src_port=5060 in_uri=sip:30125598*200@sip.voipfone.co.uk 
out_uri=sip:30125598*200@sip.voipfone.co.uk via_cnt==1"    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 501GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 0GS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER 
sip:sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bKed7898cf4d0bdb7e  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Authorization: Digest 
username="30125598*200", realm="asterisk", algorithm=MD5, 
uri="sip:sip.voipfone.co.uk", nonce="250bc905", 
response="5f1f160984712c227d093ffe9a80ae27"  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10003 REGISTER  Expires: 3888000  User-
Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 
195.189.173.12:5065 5060 543 SIP/2.0 401 Unauthorized  Via: SIP/2.0/UDP 
192.168.16.11:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bKed7898cf4d0bd
b7e  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>;tag=as018c664e  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10003 REGISTER  User-Agent: Voipfone Sip 
Network  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER  Contact: 
<sip>  WWW-Authenticate: Digest realm="asterisk", 
nonce="37fa0281"  Content-Length: 0    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 401GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 11GS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER 
sip:sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK07a3c66843378813  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Authorization: Digest 
username="30125598*200", realm="asterisk", algorithm=MD5, 
uri="sip:sip.voipfone.co.uk", nonce="37fa0281", 
response="58e8e51fbcecf0808463567e28223072"  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10004 REGISTER  Expires: 3888000  User-
Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 
195.189.173.12:5065 5060 653 SIP/2.0 200 OK  Via: SIP/2.0/UDP 
192.168.16.11:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK07a3c66843378
813  Record-Route: <sip>  Record-Route: 
<sip>  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>;tag=as018c664e  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10004 REGISTER  User-Agent: Voipfone Sip 
Network  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER  Expires: 60  Contact: 
<sip>;expires=60  Date: Sun, 03 Dec 2006 
16:28:36 GMT  Content-Length: 0    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 200GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 11GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Alan Wyatt-Jones REGISTERED for 60 
seconds;re-REGISTER in 45 secondsGS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 8 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 
SUBSCRIBE sip:30125598*200@sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bKd2598c3063f06475  From: "Alan Wyatt-Jones" 
<sip>;tag=952b1576fe27f353  To: 
<sip>;tag=dae9f5855e38e874829059f7681d766b.2a11 
Contact: <sip>  Call-ID: 
d47d9907d9a619c9@192.168.16.11  CSeq: 1002 SUBSCRIBE  User-Agent: Grandstream 
GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Event: message-summary  Expires: 3888000  Accept: application/simple-message-
summary  Content-Length: 0          GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] 195.189.173.12:5065 5060 650 SIP/2.0 
501 Not Implemented  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bKd2598c3063f06475;rport=5060;received=xx.xx.xx.xx  From: "Alan Wyatt-Jones" 
<sip>;tag=952b1576fe27f353  To: 
<sip>;tag=dae9f5855e38e874829059f7681d766b.2a11 
Call-ID: d47d9907d9a619c9@192.168.16.11  CSeq: 1002 SUBSCRIBE  Server: Sip 
EXpress router (0.8.99-dev12 (i386/linux))  Content-Length: 0  Warning: 392 
195.189.173.12:5065 "Noisy feedback tells:  pid=1404 req_src_ip=xx.xx.xx.xx 
req_src_port=5060 in_uri=sip:30125598*200@sip.voipfone.co.uk 
out_uri=sip:30125598*200@sip.voipfone.co.uk via_cnt==1"    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 501GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 0GS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER 
sip:sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK04688b7f3da25674  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Authorization: Digest 
username="30125598*200", realm="asterisk", algorithm=MD5, 
uri="sip:sip.voipfone.co.uk", nonce="37fa0281", 
response="58e8e51fbcecf0808463567e28223072"  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10005 REGISTER  Expires: 3888000  User-
Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 
195.189.173.12:5065 5060 543 SIP/2.0 401 Unauthorized  Via: SIP/2.0/UDP 
192.168.16.11:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK04688b7f3da25
674  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>;tag=as7c527af9  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10005 REGISTER  User-Agent: Voipfone Sip 
Network  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER  Contact: 
<sip>  WWW-Authenticate: Digest realm="asterisk", 
nonce="2db28868"  Content-Length: 0    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 401GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 11GS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER 
sip:sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK2a930ad9e668aa14  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Authorization: Digest 
username="30125598*200", realm="asterisk", algorithm=MD5, 
uri="sip:sip.voipfone.co.uk", nonce="2db28868", 
response="9aee8da9eb54df4904183040f858f1b0"  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10006 REGISTER  Expires: 3888000  User-
Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 
195.189.173.12:5065 5060 653 SIP/2.0 200 OK  Via: SIP/2.0/UDP 
192.168.16.11:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK2a930ad9e668a
a14  Record-Route: <sip>  Record-Route: 
<sip>  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>;tag=as7c527af9  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10006 REGISTER  User-Agent: Voipfone Sip 
Network  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER  Expires: 60  Contact: 
<sip>;expires=60  Date: Sun, 03 Dec 2006 
16:29:21 GMT  Content-Length: 0    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 200GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 11GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Alan Wyatt-Jones REGISTERED for 60 
seconds;re-REGISTER in 45 secondsGS_LOG: 




[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 8 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 
SUBSCRIBE sip:30125598*200@sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK3d7907212fb0e124  From: "Alan Wyatt-Jones" 
<sip>;tag=952b1576fe27f353  To: 
<sip>;tag=dae9f5855e38e874829059f7681d766b.2a11 
Contact: <sip>  Call-ID: 
d47d9907d9a619c9@192.168.16.11  CSeq: 1003 SUBSCRIBE  User-Agent: Grandstream 
GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Event: message-summary  Expires: 3888000  Accept: application/simple-message-
summary  Content-Length: 0          GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] 195.189.173.12:5065 5060 650 SIP/2.0 
501 Not Implemented  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK3d7907212fb0e124;rport=5060;received=81.178.45.
174  From: "Alan Wyatt-Jones" 
<sip>;tag=952b1576fe27f353  To: 
<sip>;tag=dae9f5855e38e874829059f7681d766b.2a11 
Call-ID: d47d9907d9a619c9@192.168.16.11  CSeq: 1003 SUBSCRIBE  Server: Sip 
EXpress router (0.8.99-dev12 (i386/linux))  Content-Length: 0  Warning: 392 
195.189.173.12:5065 "Noisy feedback tells:  pid=1454 req_src_ip=xx.xx.xx.xx 
req_src_port=5060 in_uri=sip:30125598*200@sip.voipfone.co.uk 
out_uri=sip:30125598*200@sip.voipfone.co.uk via_cnt==1"    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 501GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 0GS_LOG: 

--> Begin strange/bad request <--
00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To 
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER 
sip:sip.voipfone.co.uk SIP/2.0  Via: SIP/2.0/UDP 
192.168.16.11:5060;branch=z9hG4bK2a58a14f8512f1f7  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Authorization: Digest 
username="30125598*200", realm="asterisk", algorithm=MD5, 
uri="sip:sip.voipfone.co.uk", nonce="2db28868", 
response="9aee8da9eb54df4904183040f858f1b0"  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10007 REGISTER  Expires: 3888000  User-
Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: 
--> End strange/bad request <--


[00:0b:82:08:9b:04][000][FF71][0101010E] Send SIP message: 1 To
195.189.173.12:5065GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 956 REGISTER sip:sip.voipfone.co.uk 
SIP/2.0  Via: SIP/2.0/UDP 192.168.16.11:5060;branch=z9hG4bK2a58a14f8512f1f7 
From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>  Contact: 
<sip>  Authorization: Digest 
username="30125598*200", realm="asterisk", algorithm=MD5, 
uri="sip:sip.voipfone.co.uk", nonce="2db28868", 
response="9aee8da9eb54df4904183040f858f1b0"  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10007 REGISTER  Expires: 3888000  User-
Agent: Grandstream GXP2000 1.1.1.14  Max-Forwards: 70  Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0          GS_LOG: [00:0b:82:08:9b:04][000][FF71][0101010E] 
195.189.173.12:5065 5060 479 SIP/2.0 403 Forbidden  Via: SIP/2.0/UDP 
192.168.16.11:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK2a58a14f8512f
1f7  From: "Alan Wyatt-Jones" 
<sip>;tag=9f43485b063b9534  To: 
<sip>;tag=as03a94719  Call-ID: 
7ff47dc957ede49e@192.168.16.11  CSeq: 10007 REGISTER  User-Agent: Voipfone Sip 
Network  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER  Contact: 
<sip>  Content-Length: 0    GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] Received SIP message: 403GS_LOG: 
[00:0b:82:08:9b:04][000][FF71][0101010E] SIP dialog matched to channel 11
The last two messages I believe are the most significant. It seems that each message has a Sequence No. entitled 'CSeq' and this is incremented as each message is sent. Also, it appears that there are two message types 'Send SIP Message 1' [REGISTER] and 'Send SIP Message 8' [SUBSCRIBE] it appears from the response that the latter is not implemented by VoipFone as the response is quoted as 'Not Implemented'.

The 2nd from last message seems strange. It is a REGISTER with a 'CSeq' id of 10007 but the content in the debug appears short as there is no 'Received SIP Message' response as there has been in all other messages of this type.

Then, the final mesage - another REGISTER request quotes the same 'CSeq' id of 10007 which again is very strange as these sequence id's are always incremented for each send. This requests receives a 'Forbidden' response and at this point, the phone refuses to send any further requests - it just sits there unregistered and does not try again.

So, here's where I need some help from VoipFone or Grandstream - between the two, there is an obvious problem. Either the GXP-2000 'CSeq' field is using the same sequence number for two subsequent messages and the response from VoipFone causes the firmware a problem and it locks up at this point, or VoipFone are not responding to requests on occasions (hence no response in the 2nd from last message) and again, this causes the GXP-2000 firmware a problem and the CSeq ID field is not updated, then a problem occurs.

Either way one or both are at fault. I just need some help/support to push this forward.

Look forward to your responses.

Regards.

Alan Wyatt-Jones.

PS. Incidentally, the 'received=' and 'req_src_ip=' fields in the trace have been deliberately set to 'xx.xx.xx.xx' to omit my public IP address for this posting. The true IP address is recorded in the actual log.

#2
Hi Alan,

I am having exactly the same issues as you. You will see from my post that I saw very similar behaviour from my syslog. http://www.voipfoneuserforum.com/viewtopic.php?t=1763

My suspicion is that the 403 response from voipfone (to what I am not sure) is telling the phone not to try again. And it is obliging !

Unfortunately voipfone were not interested in investigating this.

Good luck. I shall be monitoring this thread for developments.

Best regards,

Davd

#3
Sorry - should have pointed out in my last post that what I _think_ is happening is the grandstream is sending out two duplicate REGISTER requests (with same branch id) and it is to this that the voipfone server is sending the forbidden response.

Wouldn't it be nice if they sent back a more forgiving rebuff ?

David

#4
Has anyone had any luck getting these phones to work with Voipfone? We have two at different sites. Other phones on the sites are fine, but these phones just seem to deregister when used with voipfone (note they seem fine with sipgate)

Yes we have the latest firmware, yes the re-register is set to 60 seconds. Yes we have tried the NAT settings on and off!

If these phones don't work with Voipfone, can anyone suggest a phone that has the same features and DOES work!!

Chanks...
...Charliep

#5
I had a Grandstream with deregistering problems and basically the Grandstream 2000 and Voipfone just don't get on.

I solved the problem by buying an Intertex SIP-aware router, as I did not want to move away from the Voipfone service. Registration is now rock solid (keeping fingers well crossed). You can also reserve bandwidth for SIP apps if you need to.

Ted

A bumbling punter with Grandstream GXP 2000 unregistering

#6
Hello to Grandstream GXP 2000 users.... I have 4 accounts operating on my GRANDSTREAM phone

1/ Gradwell............... ROCK SOLID REGISTRATION
2/ VOIPFONE account 1................ UNREGISTERING ERRATICALLY
3/ VOIPFONE account 2............... DITTO
4/ VOIPTALK ................. ROCK SOLID REGISTRATION

How can my wife be expected to understand how to RE-BOOT the new WONDER VOIPT TELEPHONE WHICH HER HUSBAND SECRETLY BOUGHT TO CHANGE HER LIFE A NEW DRESS WOULD BE MORE APPROPRIATE THAN A DAMMED TELEPHONE WHICH WON"T WORK WHEN SHE PRESSES THE ONE TOUCH DIAL KEYS.

VOIP is supposed revolutionize the telephone world... BUT my wife will put it in a dustbin if the VOIPFONE accounts keep dropping out.

Seriously VOIPFONE SUPPORT needs to buy two of these phones..... PUT THEM ON THEIR SYSTEM ON A EXTERNAL ISP ACCOUNT and LOG WHAT IS HAPPENING........... NOT ASK BUMBLING UNEDUCATED COMPUTER ILLITERATES LIKE MYSELF TO SOLVE THE PROBLEM.

THE ALTERNATIVE IS TO GIVE UP WITH VOIPFONE ACCOUNTS AND MOVE ON TO STABLE PROVIDERS. I DON'T WANT TO DO THIS!!!
John Skelton, can be contacted directly at john@inslovakia.sk or on VOIPFONE 30096850

#8
Well thanks for that John :wink:

As you can see from other posts, and from our replies to your engagingly capitalised emails to us, not everyone has problems with Grandstreams.

As it happens we don't like them and we don't sell them but do our best to help people who have them.

If you put a good router in front of them they can behave themselves but some combinations of cheap phones and cheap routers cause problems.

We conform to SIP standards; others have adjusted their network to force them to work despite themselves but it causes other problems with proper equipment so we are not going down that route.

If you wish to stay with us your options are to try a different/better router or get a proper phone.

One other thing occurs to me - we could try authenticating you via IP. If you would like to try this please email support with your IP.
Regards,

Voipfone Customer Services

iNet Telecoms Ltd (Voipfone)
Sovereign House
227 Marsh Wall
London
E14 9SD
United Kingdom

Registered number: 05168033
Vat Registration Number 858850966

Telephone: 020 7043 5555
Fax: 020 7043 5556

Web: http://www.voipfone.co.uk
Blog: http://www.voipfoneblog.co.uk
Forum: http://www.voipfoneuserforum.co.uk
Twitter: http://www.twitter.com/voipfone

#9
OK SUPPORT Lets go through this step by step.

1/ I don't have a cheap router I have a Zyxel router

2/ So far I have not had a single suggestion from VOIPFONE SUPPORT as to which parameter might not be set correctly on the GRANDSTREAM GXP 2000 phone. ALL THAT HAS BEEN STATED IS THAT VOIPFONE DON'T LIKE THE GRANDSTREAM PHONES and there is problem after problem with GRANDSTREAM phones with VOIPFONE. It is not their problem to solve. Maybe (it has been suggested) that the manufacturer is not complying with SIP protocol. It appears that GRANDSTREAM and other manufacturers don't want to talk to VOIPFONE to correct faults and the PUNTER i.e. JOHN EDWIN SKELTON in SLOVAKIA has to LUMP this problem.

3/ HOWEVER IT IS FACT not FICTION that the GRADWELL ACCOUNT has not dropped out of the GRANDSTREAM PHONE connection ONCE since I put the phone on line with them on the 4th of February 2007. ARE THEY COMPLYING WITH SIP PROTOCOL WITH THE GRADWELL ACCOUNT THAT THEY ARE OFFERING ME.

4/ I have had the VOIPTALK ACCOUNT DROP OUT TWICE since I implemented the account on the 4th of February. So again the question is are VOIPTALK complying with SIP protocol. What is this company doing differently to their VOIP provision which provides such a stable environment... I HAVE NOT HAD TO CHANGE A SINGLE PARAMETER IN THE GRADWELL OR VOIPTALK ACCOUNTS. after activating these two accounts

5/ VOIPFONE must have dropped out a HUNDRED times at least !!!! My wife can not be expected to RE BOOT any telephone repeatedly because the line has dropped out. Nor is she capable of understanding that a line has DROPPED out and the phone has to be rebooted. It is hard enough to get her to use a new telephone and press a new ONE TOUCH KEY on the Grandstream phone.

6/ 99 percent of my small expenditure on the calls to the UK and other places on this planet has been placed through VOIPFONE. OK my calls are peanuts to VOIPFONE but to me they are important and in particular INCOMING CALLS are of importance. SORRY VOIPFONE I can not buy a UK telephone number from your organisation and implement it for my clients in the UK because the reliability of the connection simply prevents me from doing this.... THIS IS FACT NOT FICTION and I AM SORRY TO HAVE TO SAY THIS.... Because I try to use VOIPFONE for all my OUTGOING CALLS. BUT this is increasingly difficult when I have to re-boot the dammed connections so often. YOU MIGHT AS WELL FACE UP TO THIS FACT... I AM NOT DOING ANYTHING AT MY END THAT CAUSES THIS PROBLEM TO BE SPECIFIC TO VOIPFONE as far as I am aware. !!!!!!

OK this has got the problem off my chest. I am sick and tired of trying to get a stable connection to VOIPFONE accounts. FACT not FICTION. John Edwin Skelton in exile in Slovakia john@inslovakia.sk
John Skelton, can be contacted directly at john@inslovakia.sk or on VOIPFONE 30096850

#10
3 minutes after I posted the last notice... I check my status with GRANDSTEAM GXP 2000

Account 1 GRADWELL OK as always

Account 2 VOIPFONE NOT REGISTERED WELL IT WAS 3 minutes ago

Account 3 VOIPFONE Now registered WELL IT WAS NOT 8 munutes ago

account 4 VOIPTALK OK as it has been for 99.95 percent of the time since the 4th of February 2007

WHAT AM I AS A BUMBLING AMATEUR TO DO ABOUT THIS SITUATION !!!!

John Edwin in despair.
John Skelton, can be contacted directly at john@inslovakia.sk or on VOIPFONE 30096850

#11
Like I say, would you like to try IP authorisation? If so please email support.

And please press Caps Lock.
Regards,

Voipfone Customer Services

iNet Telecoms Ltd (Voipfone)
Sovereign House
227 Marsh Wall
London
E14 9SD
United Kingdom

Registered number: 05168033
Vat Registration Number 858850966

Telephone: 020 7043 5555
Fax: 020 7043 5556

Web: http://www.voipfone.co.uk
Blog: http://www.voipfoneblog.co.uk
Forum: http://www.voipfoneuserforum.co.uk
Twitter: http://www.twitter.com/voipfone

STABLE AT LAST

#12
With reference to my previous posts with regard the unstable connection to Voipfone accounts when using a Grandstream GXP 2000

OK the situation appears to have finally stabilised over the last 48 hours.

My actions have been as follows.

a/ On my ZyXel router 600 series open on the router NAT all ports to a fixed IP address. This is the fixed IP of the Grandstream GXP 2000

b/ Send the nominally static IP address of the ZyXel router to support as they requested.

Fingers crossed>>> the Voipfone accounts are not longer dropping out in the random way they were. If support has been logging my lines then the only time when there has been a drop out is when I have manually removed the phone to take to a potential client in Slovakia or when I have forced a re-boot because of changing an unrelated parameter to the line connection on the Grandstream web browser>>>>> FIngers crossed

Thanks to support if using the supplied nominally static IP as the Voipfone end of the service has cured this problem. John Skelton in SLOVAKIA
John Skelton, can be contacted directly at john@inslovakia.sk or on VOIPFONE 30096850

A STATEMENT OF FACT re Grandstream GXP 2000 & registrati

#13
Re drop out problems with voipfone accounts and Grandstream GXP 2000

1/ This problem was causing my company acute problems, with two accounts registered with VOIPFONE and my business relying on these accounts in the Slovak Republic.

2/ Chasing the problem by changing the parameters on the GXP 2000 web browser did not produce any tangible results in stability. The lines continued to drop out rapidly and on an irregular time pattern.

3/ Contact with Grandstream directly produced an almost immediate response from their technical department. Benny Rozeki made useful and concise comments to me.

4/ I passed Grandstream details of 'support' in voipfone on 22/02/2007.

5/ My accounts continued to be erratically registering on the two accounts causing concern until the 7th March. Suddenly one account became stable.... The second account continued to drop out erratically until the 8th of March....

I had not changed a single parameter on the settings to these accounts for voipfone accounts for several days prior to the 7th March 2007.

I hope that the current stable situation will continue. The GRANDSTREAM GXP 2000 has such a super LCD display and easy to use one touch dialing for my wife..... Peace has now returned to the Skelton household. The GXP 2000 phone is no longer under threat of being dumped in the dustbin....

Anyone else noticed an improvement with their Grandstream phones lately ??? john@inslovakia.sk
_________________
John Skelton, can be contacted directly at john@inslovakia.sk or on VOIPFONE 30096850

Link to Grandstream FIRMWARE UP LOAD PAGE

#14
To Help anyone with UPLOADING the latest FIRMWARE for GRANDSTREAM GXP 2000 phones PLEASE CLICK ON THE LINK BELOW.

GRANDSTREAM also state that another FIRMWARE UPGRADE is currently in BETA TEST MODE.

When this is released to the general user (Like me a 59 year old bumbling amateur) I will post again OK

MEANWHILE I can recommend that this upgrade is worth installing on the GPX 2000 phone from Grandstream.


Hope that this link will help other users.

http://www.grandstream.com/firmware.html

John Skelton Yorkshire man in exile in the Slovak Republic....
John Skelton, can be contacted directly at john@inslovakia.sk or on VOIPFONE 30096850

#15
It has to be said I have noticed that the latest firmware seems to have made the connection a LOT more stable. It is still by no means 100%, but I can usually get a week before I have to reboot the phone!

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