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Kanuri, Seshu

2004-Jul-19 14:03 UTC

[Asterisk-Users] Affordable SIP Phone - Stiil a Myth?

Folks!

This is to let all of you know that I am making D'Link make an all out
effort to make D'Link Phone DPH80 and DPH100 work with Asterisk. I have
provided the Asterisk Platform to D'Link's R&D Division located in
Goa, India, where their IP phone's SIP Bios is undergoing modifications
based on my recommendations/suggestions. I have also provided the test bed &
manpower for the tests.

After 2 months of sleepless nights, the phones are finally working. Some of the
initial problems we found were as under:

1)These phones auto Un-register dfrom Asterisk after 30 seconds
2)No provison to give the NAT IP address for STUN Server
3)Line rings after connect to Asterisk Extension but the Phone does not pickup
up the line

...etc. etc.

This phone would soon be available in India(by next week or so, first branded
asr "Netweb Phone") and USA in the next couple of months as
"NetwebPhone" and could be priced around $65/-(tentative)

See the communication from D'Link below.

Seshu Kanuri


----- Original Message ----- 
From: Mandar Pise (Netweb India Ltd)
To: Abhijit M ( Voip Dept.) 
Sent: Monday, July 19, 2004 1:51 PM
Subject: Re: D-link ( DPH-80 ),Sip Firmware Upgrade .


Dear Mr. Abhijit,
 
As per our teleconversation with you, I am forwarding you the sip server logs
attached along with this email.
Thank you,

Regards,
Mandar Pise

"Abhijit M ( Voip Dept.)" <amalankar@dlink.co.in> wrote: 
Dear Raghvendra / Mandar , 

Forwarding to you the Dph-80 , upgrade of firmware which now works with your Sip
Server ( 67.109.153.236).
Please remember to factory reset once the upgrade through TFTP server is over
before any further configuration.
Now the user name , passwords for accessing the phones through web are kept
blank.
Factory reset can be done through keys using *789*# .
Also enable log server in Network settings with your sip server ip (
67.109.153.236).
Please inform us what port you are using for log server .
Also in sip configuration , disable auto attendant , enable vad , keep
user=phone enabled .
Always click submit on every page you are configuring and at last click = save
and restart .
Now the phones will only be accessible by there new ips , and should give a dial
tone .

I have tested this in-between two DPH-80s and from Mr.Mandars side at pune
in-between soft phone to our hard phone . with G.729 and G.711 enabled , its
working perfectly well and getting registered on your server .

Mr.Mandar please forward me all the server logs for today between 2006 &
2008 nos that I have configured at my end .
Thanks & regards,

Abhijit M.
VoIP Dept.
D-Link India Ltd.
Mumbai.
Phone: +91-22-2652 6696/56902210, Ext-194
Fax     : +91-22-26528914

------------------------------------------------------------------------------------
Log
------------------------------------------------------------------------------------
Jul 15 01:45:42 WARNING[-151058624]: Unable to open /dev/dsp: No such 
device
Jul 15 01:45:42 WARNING[-151058624]: Invalid localnet keyword: 
192.168.0.0/255.255.0.0
Jul 15 01:45:52 WARNING[-151058624]: Invalid localnet keyword: 
10.0.0.0/255.0.0.0
Jul 15 01:46:02 WARNING[-151058624]: Invalid localnet keyword: 
172.16.0.0/12
Jul 15 01:46:13 WARNING[-151058624]: Invalid localnet keyword: 
169.254.0.0/255.255.0.0
Jul 15 01:46:13 VERBOSE[-151058624]:     -- SIP Seeding '8612312342' at 
8612312342@61.1.106.200:5061 for 1800
Jul 15 01:46:13 WARNING[-151058624]: Ignoring port for now
Jul 15 01:46:13 WARNING[-151058624]: MySQL database sock file not 
specified.  Using default
Jul 15 01:46:20 WARNING[-203732048]: Maximum retries exceeded on call 
184e51060245db8517524fea1097a652@67.109.153.236 for seqno 102 (Request)
Jul 15 01:55:50 WARNING[-288244816]: MySQL database sock file not 
specified.  Using default
Jul 15 01:55:50 WARNING[-288244816]: Ignoring port for now
Jul 15 01:55:50 NOTICE[-288244816]: Removed default indication country 
'us'
Jul 15 01:56:00 WARNING[-203732048]: Invalid localnet keyword: 
192.168.0.0/255.255.0.0
Jul 15 01:56:10 WARNING[-203732048]: Invalid localnet keyword: 
10.0.0.0/255.0.0.0
Jul 15 01:56:20 WARNING[-203732048]: Invalid localnet keyword: 
172.16.0.0/12
Jul 15 01:56:30 WARNING[-203732048]: Invalid localnet keyword: 
169.254.0.0/255.255.0.0
Jul 15 01:56:31 WARNING[-298812496]: No entry in voicemail config file 
for '8612312344'
Jul 15 01:56:36 WARNING[-203732048]: Maximum retries exceeded on call 
783DD50B-BF1A-4E03-9D3F-1912F8A5AF90@61.1.106.200 for seqno 57664 
(Response)
Jul 15 02:00:05 WARNING[-288244816]: No entry in voicemail config file 
for '8612312344'
Jul 15 02:02:42 WARNING[-288244816]: No entry in voicemail config file 
for '8612312344'
Jul 15 02:04:01 WARNING[-288244816]: No entry in voicemail config file 
for '8612312344'
Jul 15 02:07:20 WARNING[-203732048]: Maximum retries exceeded on call 
357cc336111ccf7b5b536cb1649b8b1c@67.109.153.236 for seqno 102 (Request)
Jul 15 02:07:20 WARNING[-288244816]: No entry in voicemail config file 
for '8612312344'
Jul 15 02:07:26 WARNING[-203732048]: Maximum retries exceeded on call 
357cc336111ccf7b5b536cb1649b8b1c@67.109.153.236 for seqno 102 (Request)
Jul 15 02:09:39 WARNING[-203732048]: Maximum retries exceeded on call 
52de2b95153f7bf17d84c8ef4bb26743@67.109.153.236 for seqno 102 (Request)
Jul 15 02:09:39 WARNING[-288244816]: No entry in voicemail config file 
for '8612312344'
Jul 15 02:09:45 WARNING[-203732048]: Maximum retries exceeded on call 
52de2b95153f7bf17d84c8ef4bb26743@67.109.153.236 for seqno 102 (Request)
Jul 15 02:25:44 VERBOSE[-203732048]: 

Sip read: 
REGISTER sip:67.109.153.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>
Call-ID: 16838@192.168.100.108
Contact: 
<sip:2009@192.168.100.108:5060;user=phone>,<sip:2009@192.168.100.108:5060>
CSeq: 6065 REGISTER
Max-Forwards: 70
Content-Length: 0
User-Agent: D-Link DPH80 (TW-3.200)
Expires: 0


Jul 15 02:25:44 VERBOSE[-203732048]: 11 headers, 0 lines
Jul 15 02:25:44 VERBOSE[-203732048]: Using latest request as basis 
request
Jul 15 02:25:44 VERBOSE[-203732048]: Sending to 192.168.100.108 : 5060 
(non-NAT)
Jul 15 02:25:44 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6065 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:25:44 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6065 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Proxy-Authenticate: Digest realm="asterisk",
nonce="6bfd582a"
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:25:44 VERBOSE[-203732048]: Retransmitting #3 (NAT):
NOTIFY sip:8612312342@61.1.106.200:5061 SIP/2.0
Via: SIP/2.0/UDP 67.109.153.236:5060;branch=z9hG4bK6870750e
From: "asterisk" <sip:asterisk@67.109.153.236>;tag=as27b26817
To: <sip:8612312342@61.1.106.200:5061>
Contact: <sip:asterisk@67.109.153.236>
Call-ID: 41f2abc40da35ba877508b704ae90b7d@67.109.153.236
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0
0
 to 61.1.106.200:5061
Jul 15 02:25:45 VERBOSE[-203732048]: 

Sip read: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.109.153.236:5060;branch=z9hG4bK6870750e
From: "asterisk" <sip:asterisk@67.109.153.236>;tag=as27b26817
To: <sip:8612312342@61.1.106.200:5060>;tag=as27b26817
Contact: <sip:8612312342@61.1.106.200:5061>
Call-ID: 41f2abc40da35ba877508b704ae90b7d@67.109.153.236
CSeq: 102 NOTIFY
User-Agent: X-PRO build 1082
Content-Length: 0


Jul 15 02:25:45 VERBOSE[-203732048]: 9 headers, 0 lines
Jul 15 02:25:46 VERBOSE[-203732048]: 

Sip read: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.109.153.236:5060;branch=z9hG4bK6870750e
From: "asterisk" <sip:asterisk@67.109.153.236>;tag=as27b26817
To: <sip:8612312342@61.1.106.200:5060>;tag=as27b26817
Contact: <sip:8612312342@61.1.106.200:5061>
Call-ID: 41f2abc40da35ba877508b704ae90b7d@67.109.153.236
CSeq: 102 NOTIFY
User-Agent: X-PRO build 1082
Content-Length: 0


Jul 15 02:25:46 VERBOSE[-203732048]: 9 headers, 0 lines
Jul 15 02:25:47 VERBOSE[-203732048]: 

Sip read: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.109.153.236:5060;branch=z9hG4bK6870750e
From: "asterisk" <sip:asterisk@67.109.153.236>;tag=as27b26817
To: <sip:8612312342@61.1.106.200:5060>;tag=as27b26817
Contact: <sip:8612312342@61.1.106.200:5061>
Call-ID: 41f2abc40da35ba877508b704ae90b7d@67.109.153.236
CSeq: 102 NOTIFY
User-Agent: X-PRO build 1082
Content-Length: 0


Jul 15 02:25:47 VERBOSE[-203732048]: 9 headers, 0 lines
Jul 15 02:25:54 VERBOSE[-203732048]: 

Sip read: 
REGISTER sip:67.109.153.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>
Call-ID: 16838@192.168.100.108
Contact: 
<sip:2009@192.168.100.108:5060;user=phone>,<sip:2009@192.168.100.108:5060>
CSeq: 6066 REGISTER
Max-Forwards: 70
Content-Length: 0
User-Agent: D-Link DPH80 (TW-3.200)
Expires: 0


Jul 15 02:25:54 VERBOSE[-203732048]: 11 headers, 0 lines
Jul 15 02:25:54 VERBOSE[-203732048]: Using latest request as basis 
request
Jul 15 02:25:54 VERBOSE[-203732048]: Sending to 192.168.100.108 : 5060 
(non-NAT)
Jul 15 02:25:54 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6066 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:25:54 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6066 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Proxy-Authenticate: Digest realm="asterisk",
nonce="3d2d1722"
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:25:59 VERBOSE[-203732048]: 

Sip read: 
REGISTER sip:67.109.153.236 SIP/2.0
Via: SIP/2.0/UDP 
61.1.106.200:5061;rport;branch=z9hG4bK6C03F7B7F6DE47C7ACA4EC0CA356A778
From: mandar <sip:8612312342@67.109.153.236>
To: mandar <sip:8612312342@67.109.153.236>
Contact: "mandar" <sip:8612312342@61.1.106.200:5061>
Call-ID: 9BDF5A691EB246C88E9A39B01947F30D@67.109.153.236
CSeq: 23666 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-PRO build 1082
Content-Length: 0


Jul 15 02:25:59 VERBOSE[-203732048]: 11 headers, 0 lines
Jul 15 02:25:59 VERBOSE[-203732048]: Using latest request as basis 
request
Jul 15 02:25:59 VERBOSE[-203732048]: Sending to 61.1.106.200 : 5061 
(non-NAT)
Jul 15 02:25:59 VERBOSE[-203732048]: Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
61.1.106.200:5061;rport;branch=z9hG4bK6C03F7B7F6DE47C7ACA4EC0CA356A778;received=61.1.106.200
From: mandar <sip:8612312342@67.109.153.236>
To: mandar <sip:8612312342@67.109.153.236>;tag=as57353929
Call-ID: 9BDF5A691EB246C88E9A39B01947F30D@67.109.153.236
CSeq: 23666 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8612312342@67.109.153.236>
Content-Length: 0


 to 61.1.106.200:5061
Jul 15 02:25:59 VERBOSE[-203732048]: Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
61.1.106.200:5061;rport;branch=z9hG4bK6C03F7B7F6DE47C7ACA4EC0CA356A778;received=61.1.106.200
From: mandar <sip:8612312342@67.109.153.236>
To: mandar <sip:8612312342@67.109.153.236>;tag=as57353929
Call-ID: 9BDF5A691EB246C88E9A39B01947F30D@67.109.153.236
CSeq: 23666 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8612312342@67.109.153.236>
Proxy-Authenticate: Digest realm="asterisk",
nonce="1bfdf202"
Content-Length: 0


 to 61.1.106.200:5061
Jul 15 02:26:00 VERBOSE[-203732048]: 

Sip read: 
REGISTER sip:67.109.153.236 SIP/2.0
Via: SIP/2.0/UDP 
61.1.106.200:5061;rport;branch=z9hG4bK6C03F7B7F6DE47C7ACA4EC0CA356A778
From: mandar <sip:8612312342@67.109.153.236>
To: mandar <sip:8612312342@67.109.153.236>
Contact: "mandar" <sip:8612312342@61.1.106.200:5061>
Call-ID: 9BDF5A691EB246C88E9A39B01947F30D@67.109.153.236
CSeq: 23666 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-PRO build 1082
Content-Length: 0


Jul 15 02:26:00 VERBOSE[-203732048]: 11 headers, 0 lines
Jul 15 02:26:00 VERBOSE[-203732048]: Using latest request as basis 
request
Jul 15 02:26:00 VERBOSE[-203732048]: Sending to 61.1.106.200 : 5061 
(NAT)
Jul 15 02:26:00 VERBOSE[-203732048]: Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
61.1.106.200:5061;rport;branch=z9hG4bK6C03F7B7F6DE47C7ACA4EC0CA356A778;received=61.1.106.200
From: mandar <sip:8612312342@67.109.153.236>
To: mandar <sip:8612312342@67.109.153.236>;tag=as57353929
Call-ID: 9BDF5A691EB246C88E9A39B01947F30D@67.109.153.236
CSeq: 23666 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8612312342@67.109.153.236>
Content-Length: 0


 to 61.1.106.200:5061
Jul 15 02:26:00 VERBOSE[-203732048]: Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
61.1.106.200:5061;rport;branch=z9hG4bK6C03F7B7F6DE47C7ACA4EC0CA356A778;received=61.1.106.200
From: mandar <sip:8612312342@67.109.153.236>
To: mandar <sip:8612312342@67.109.153.236>;tag=as57353929
Call-ID: 9BDF5A691EB246C88E9A39B01947F30D@67.109.153.236
CSeq: 23666 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8612312342@67.109.153.236>
Proxy-Authenticate: Digest realm="asterisk",
nonce="1bfdf202"
Content-Length: 0


 to 61.1.106.200:5061
Jul 15 02:26:04 VERBOSE[-203732048]: 

Sip read: 
REGISTER sip:67.109.153.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>
Call-ID: 16838@192.168.100.108
Contact: 
<sip:2009@192.168.100.108:5060;user=phone>,<sip:2009@192.168.100.108:5060>
CSeq: 6067 REGISTER
Max-Forwards: 70
Content-Length: 0
User-Agent: D-Link DPH80 (TW-3.200)
Expires: 0


Jul 15 02:26:04 VERBOSE[-203732048]: 11 headers, 0 lines
Jul 15 02:26:04 VERBOSE[-203732048]: Using latest request as basis 
request
Jul 15 02:26:04 VERBOSE[-203732048]: Sending to 192.168.100.108 : 5060 
(non-NAT)
Jul 15 02:26:04 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6067 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:26:04 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6067 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Proxy-Authenticate: Digest realm="asterisk",
nonce="3a55c3bc"
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:26:14 VERBOSE[-203732048]: 

Sip read: 
REGISTER sip:67.109.153.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>
Call-ID: 16838@192.168.100.108
Contact: 
<sip:2009@192.168.100.108:5060;user=phone>,<sip:2009@192.168.100.108:5060>
CSeq: 6068 REGISTER
Max-Forwards: 70
Content-Length: 0
User-Agent: D-Link DPH80 (TW-3.200)
Expires: 0


Jul 15 02:26:14 VERBOSE[-203732048]: 11 headers, 0 lines
Jul 15 02:26:14 VERBOSE[-203732048]: Using latest request as basis 
request
Jul 15 02:26:14 VERBOSE[-203732048]: Sending to 192.168.100.108 : 5060 
(non-NAT)
Jul 15 02:26:14 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6068 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:26:14 VERBOSE[-203732048]: Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>;tag=as5d00ba9d
Call-ID: 16838@192.168.100.108
CSeq: 6068 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2009@67.109.153.236>
Proxy-Authenticate: Digest realm="asterisk",
nonce="4a43ef31"
Content-Length: 0


 to 192.168.100.108:5060
Jul 15 02:26:21 VERBOSE[-203732048]: 

Sip read: 


Jul 15 02:26:21 VERBOSE[-203732048]: 0 headers, 0 lines
Jul 15 02:26:24 VERBOSE[-203732048]: 

Sip read: 
REGISTER sip:67.109.153.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.108:5060
From: 2009 <sip:2009@67.109.153.236>
To: 2009 <sip:2009@67.109.153.236>
Call-ID: 16838@192.168.100.108
Contact: 
<sip:2009@192.168.100.108:5060;user=phone>,<sip:2009@192.168.100.108:5060>
CSeq: 6069 REGISTER
Max-Forwards: 70
Content-Length: 0
User-Agent: D-Link DPH80 (TW-3.200)
Expires: 0

Philipp von Klitzing

2004-Jul-19 15:21 UTC

head link

[Asterisk-Users] Affordable SIP Phone - Stiil a Myth?

Hi!
> Also in sip configuration , disable auto attendant , enable vad
vad is not supported by Asterisk

Cheers, Philipp

Kanuri, Seshu

2004-Jul-20 08:08 UTC

head link

[Asterisk-Users] Affordable SIP Phone - Stiil a Myth?

D'Link has thrown one more spanner into my works.

Here is the clarification I have got about the STUN functionality. 
We need to buy their SIP server, if we have to make the D'Link Phones 
work transparently or Asterisk Server should have a Stun Server running on the
same IP.

Seshu Kanuri

From: "G Rao" <grkalaga@dlink.co.in>  Add to Address Book 
To: "Seshu Kanuri" <seshukanuri@yahoo.com> 
CC: mandarpise@yahoo.com 
Subject: Re: STUN server settings 
Date: Tue, 20 Jul 2004 20:30:26 +0530 

    
Dear Mr. Seshu Kanuri,

Thanks for your mail.
1. D-Link IP Phones do not have any settings for the STUN server.
2. D-Link (SIP + H.323) Server has in-built STUN functionality support.
3. If one uses this D-Link Server the D-Link IP Phones (DPH-80) can work
behind a NAT also.
4. In general if the Stun functionality, if in-built into the SIP server,
then the end-devices do need to have any STUN support.
5. If D-Link end devices (IP Phones) if used with any other SIP server which
do not have in-built STUN support, then they may not work behind NAT.
I hope I am clear.

Thanks / Regards,
Rao



KVSSS GUNNESWARA RAO
D-Link (India) Limited.
Phone: +91 22 2650 6271
Mobile: +91 98212 18057
----- Original Message ----- 
From: "Seshu Kanuri" <seshukanuri@yahoo.com>
To: "G Rao" <grkalaga@dlink.co.in>
Cc: <mandarpise@yahoo.com>
Sent: Tuesday, July 20, 2004 7:09 PM
Subject: Re: STUN server settings

> Dr Rao Garu,
>
> I need a lttle more clarification on this.
>
> 1) Do we or dont we need a Stun Server running on our SIP server IP?
> 2) Does the D'Link Phones use D'Link's STUN server by default?
> /*
> 1. D-Link India has one SIP + H.323 server which has the in-built STUN
server>    support.
> 2. So the end-devices (IP Phones) do not have any separate stun suport.
> */
>
> 3) Point 2 - Does this mean that D'Link Phones by default dont have
Stun
> Support and will not work from NAT?
>
> Please clarify.
>
> Thank You
>
> Seshu Kanuri
>
>
> --- G Rao <grkalaga@dlink.co.in> wrote:
> > Dear Mandar,
> >
> > You are right.
> >
> > 1. D-Link India has one SIP + H.323 server which has the in-built STUN
server> > support.
> > 2. So the end-devices (IP Phones) do not have any separate stun
suport.
> >
> > Thanks / Regards,
> > Rao
> >
> >
> > KVSSS GUNNESWARA RAO
> > D-Link (India) Limited.
> > Phone: +91 22 2650 6271
> > Mobile: +91 98212 18057
> >   ----- Original Message ----- 
> >   From: Mandar Pise
> >   To: grkalaga@dlink.co.in
> >   Cc: Seshu Kanuri
> >   Sent: Tuesday, July 20, 2004 4:08 PM
> >   Subject: STUN server settings
> >
> >
> >   Dear Mr. Rao,
> >
> >
> >
> >   This is in reference to our telecon few minutes ago; I am listing
the
> > setting that I understood.
> >
> >
> >
> >   The STUN server must be run on the SIP server IP address to resolve
NAT> > issue of IP phone. There is no necessity of separate field for STUN
server> > address in IP phone.
> >
> >
> >
> >   Kindly correct me if I misunderstood something so I can convey the
same to> > our US technicians.
> >
> >
> >
> >
> >
> >   Thanks & Regards,
> >
> >   Mandar Pise
> >
> >
> >

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