[Asterisk-Users] IP SIP Phone/2.0.6

[Richard Smith]

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 at pbx.sytes.net>;tag=22d32002

Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5 at 192.168.1.3

CSeq: 2230 REGISTER

Expires: 30

User-Agent: IP SIP Phone/2.0.6

Max-Forwards: 70

Contact: <sip:119 at 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 at pbx.sytes.net>;tag=22d32002

To: "RN SYSTEMS LTD" <sip:119 at pbx.sytes.net>

Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5 at 192.168.1.3

CSeq: 2230 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY

Contact: <sip:119 at 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 at pbx.sytes.net>;tag=22d32002

To: "RN SYSTEMS LTD" <sip:119 at pbx.sytes.net>;tag=as5fc91493

Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5 at 192.168.1.3

CSeq: 2230 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY

Contact: <sip:119 at 82.35.xxx.xxx>

WWW-Authenticate: Digest realm="asterisk", nonce="3eaaf09a"

Content-Length: 0

 

 

---

Scheduling destruction of call '0a4132dc-44a3276f-8d5e4f44-bc07bcd5 at 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 at pbx.sytes.net>

From: "RN SYSTEMS LTD" <sip:119 at pbx.sytes.net>;tag=22d32002

Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5 at 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 at 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 at pbx.sytes.net>;tag=22d32002

To: "RN SYSTEMS LTD" <sip:119 at pbx.sytes.net>

Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5 at 192.168.1.3

CSeq: 2231 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY

Contact: <sip:119 at 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 at pbx.sytes.net>;tag=22d32002

To: "RN SYSTEMS LTD" <sip:119 at pbx.sytes.net>;tag=as5fc91493

Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5 at 192.168.1.3

CSeq: 2231 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY

Expires: 30

Contact: <sip:119 at 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 at 192.168.1.3' in 15000 ms

asterisk1*CLI>

<-- SIP read from 82.35.xxx.23:33344:

SUBSCRIBE sip:*97 at 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 at pbx.sytes.net>;tag=as11253045

From: "RN SYSTEMS LTD" <sip:119 at pbx.sytes.net>;tag=231d2266

Call-ID: 794d560e-b9292356-691ac354-297738a3 at 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 at 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 at pbx.sytes.net>;tag=231d2266

To: "MB" <sip:*97 at pbx.sytes.net>;tag=as11253045

Call-ID: 794d560e-b9292356-691ac354-297738a3 at 82.35.xxx.23

CSeq: 8050 SUBSCRIBE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY

Contact: <sip:*97 at 82.35.xxx.xxx>

Content-Length: 0

 

 

---

Destroying call '794d560e-b9292356-691ac354-297738a3 at 82.35.xxx.23'



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