Dear all,
I know, you get what you pay for. I bought an IP SIP Phone/2.0.6 from safe.com
(?55) and the basic functionality is fine.
The problem is when it tries to re-register it hangs for a minute or so and you
can not dial nor receive any calls. It also
has a registration button which causes the phone to do the same once pressed.
This however this does not occur when the phone registers direct with a VOIP
provider.
I was wondering whether anybody else has come across this kind of problem
before.
Thanks for your anticipated help.
Cheers,
Richard
Here is a snippet from the debug log;
asterisk1*CLI>
<-- SIP read from 82.35.xxx.23:33344:
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK21852074
To: "RN SYSTEMS LTD" <sip:119@ pbx.sytes.net >
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=22d32002
Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3
CSeq: 2230 REGISTER
Expires: 30
User-Agent: IP SIP Phone/2.0.6
Max-Forwards: 70
Contact: <sip:119@82.35.xxx.23:33344>
--- (11 headers 0 lines)---
Using latest request as basis request
Sending to 82.35.xxx.23 : 33344 (non-NAT)
Transmitting (no NAT) to 82.35.xxx.23:33344:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK21852074
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=22d32002
To: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>
Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3
CSeq: 2230 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:119@82.35.xxx.23>
ontent-Length: 0
---
Transmitting (no NAT) to 82.35.xxx.23:33344:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK21852074
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=22d32002
To: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=as5fc91493
Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3
CSeq: 2230 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:119@82.35.xxx.xxx>
WWW-Authenticate: Digest realm="asterisk", nonce="3eaaf09a"
Content-Length: 0
---
Scheduling destruction of call
'0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3' in 15000 ms
asterisk1*CLI>
<-- SIP read from 82.35.xxx.23:33344:
REGISTER sip:pbx.sytes.net SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK219a2337
To: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=22d32002
Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3
CSeq: 2231 REGISTER
Expires: 30
User-Agent: IP SIP Phone/2.0.6
Max-Forwards: 70
Authorization: Digest nonce="3eaaf09a", username="119",
realm="asterisk", uri="sip:pbx.sytes.net",
response="9d12fb8ddbfc05b6f9e0e10c074fcf89"
P-IPRAuth: asterisk
Contact: <sip:119@82.35.xxx.23:33344>
--- (13 headers 0 lines)---
Using latest request as basis request
Sending to 82.35.xxx.23 : 33344 (non-NAT)
Transmitting (no NAT) to 82.35.xxx.23:33344:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK219a2337
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=22d32002
To: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>
Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3
CSeq: 2231 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:119@82.35.xxx.xxx>
Content-Length: 0
---
Transmitting (no NAT) to 82.35.xxx.23:33344:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.35.xxxx.23:33344;branch=z9hG4bK219a2337
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=22d32002
To: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=as5fc91493
Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3
CSeq: 2231 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Expires: 30
Contact: <sip:119@82.35.xxx.23:33344>;expires=30
Date: Sun, 22 Jan 2006 22:38:18 GMT
Content-Length: 0
---
Scheduling destruction of call
'0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3' in 15000 ms
asterisk1*CLI>
<-- SIP read from 82.35.xxx.23:33344:
SUBSCRIBE sip:*97@pbx.sytes.net SIP/2.0
Content-Length: 0
Date: Mon, 22 Jan 2006 22:38:29 GMT
Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK24892096
To: "MB" <sip:*97@pbx.sytes.net>;tag=as11253045
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=231d2266
Call-ID: 794d560e-b9292356-691ac354-297738a3@82.35.xxx.23
Event: message-summary
CSeq: 8050 SUBSCRIBE
Expires: 0
User-Agent: IP SIP Phone/2.0.6
Max-Forwards: 70
Accept: application/simple-message-summary
Contact: <sip:119@82.35.xxx.23:33344>
--- (14 headers 0 lines)---
Using latest SUBSCRIBE request as basis request
Sending to 82.35.xxx.23 : 33344 (non-NAT)
Found peer '119'
Looking for *97 in from-internal
Transmitting (no NAT) to 82.35.xxx.23:33344:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK24892096
From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=231d2266
To: "MB" <sip:*97@pbx.sytes.net>;tag=as11253045
Call-ID: 794d560e-b9292356-691ac354-297738a3@82.35.xxx.23
CSeq: 8050 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:*97@82.35.xxx.xxx>
Content-Length: 0
---
Destroying call '794d560e-b9292356-691ac354-297738a3@82.35.xxx.23'
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