[Asterisk-Dev] Sip Expert Help REquested

Wed Oct 27 08:59:44 MST 2004

Yes, you need to include a sample of your config files please.

Sip.conf
Extension.conf

Please remove any private information.

>From the sip debug messages:

Asterisk is receiving the message but not acting on them.

Do you have autocreatepeer=yes in the config file?. Sip.conf
http://www.voip-info.org/wiki-Asterisk+sip+autocreatepeer

Or 

Do you have the phone user set up in sip.conf?
http://www.voip-info.org/wiki-Asterisk+config+sip.conf

If there is no user or autocreatepeer is not turned on then Asterisk
will not allow the phone to register.

If the phone is registering then you will get a ACK 200.

Make sure you don't have the MD5 turned on in the phone or the sip.conf
until you get the phone to register in the open. Then look at md5secret=
in the voip wiki
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20md5secret

Race Vanderdecken

Asterisk AT VanDerDecKEN period Com

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Ronald
Hartmann
Sent: 26 October 2004 14:49
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] Sip Expert Help REquested

Good Day list,

First off, I want to apologize if this is outside the lists purpose.

I have posted to the users list and have had no response from them over
the last 5 days, 
After much digging, I have come up with nothing but my observations.

I have purchased a Video Phone with SIP capabilities, however I do not
think that the SIP 
Solution is not complete, If my observations are correct, I think that
the phone does not have a user
Agent which responds to Asterisk closing the loop on handshaking and
making both sides aware of the other.

Here is my info to date.

Any links for additional reading or advise would be greatly appreciated.

WookSung TelephoSee 2000 Help needed.

Can not get the phone to register with asterisk.  I am not sure what the
problem is at this point.

I have the setup of the phone as:

Server1 192.168.3.1
Port1: 5060

Display: TelephoSee
URI: <blank>  
Userid: 2205
Password: "password"

Following is the debug.   Any assistance would be helpful...

pc-11*CLI> sip debug
SIP Debugging Enabled
pc-11*CLI>
pc-11*CLI>
pc-11*CLI>
pc-11*CLI>
pc-11*CLI>
pc-11*CLI>
pc-11*CLI>
pc-11*CLI>
Destroying call '043194642e7f2779c77bd47b885ca423 at 192.168.3.23'
pc-11*CLI>

Sip read:
REGISTER sip:192.168.3.11:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137
To: <sip:@192.168.3.11:5060;user=phone>
From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204
Call-ID: 579bb2735be96d498b77c70cf8c00706 at 192.168.3.23
CSeq: 1 REGISTER
Max-Forwards: 70
Expires: 3600
Contact: <sip:@192.168.3.23:5060;user=phone>
Content-Length: 0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.3.23 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137
From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204
To: <sip:@192.168.3.11:5060;user=phone>;tag=as5975a76c
Call-ID: 579bb2735be96d498b77c70cf8c00706 at 192.168.3.23
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:192.168.3.11>
Content-Length: 0

 to 192.168.3.23:5060
Scheduling destruction of call
'579bb2735be96d498b77c70cf8c00706 at 192.168.3.23' in 15000 ms
pc-11*CLI>

Sip read:
REGISTER sip:192.168.3.11 SIP/2.0
Via: SIP/2.0/UDP
192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137
To: <sip:@192.168.3.11:5060;user=phone>
From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204
Call-ID: 579bb2735be96d498b77c70cf8c00706 at 192.168.3.23
CSeq: 2 REGISTER
Max-Forwards: 70
Expires: 0
Contact: <sip:@192.168.3.23:5060;user=phone>
Content-Length: 0

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.3.23 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137
From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204
To: <sip:@192.168.3.11:5060;user=phone>;tag=as5975a76c
Call-ID: 579bb2735be96d498b77c70cf8c00706 at 192.168.3.23
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:192.168.3.11>
Content-Length: 0

 to 192.168.3.23:5060
Scheduling destruction of call
'579bb2735be96d498b77c70cf8c00706 at 192.168.3.23' in 15000 ms
Destroying call '579bb2735be96d498b77c70cf8c00706 at 192.168.3.23'

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