[Asterisk-Dev] Sip Expert Help REquested
Steve Totaro
asterisk at totarotechnologies.com
Tue Oct 26 12:48:10 MST 2004
is there something to check on the phone config that says user=phone? grandstream does this when that box is checked.
Thanks,
Steve Totaro
stotaro at totarotechnologies.com
www.totarotechnologies.com
----- Original Message -----
From: Ronald Hartmann
To: asterisk-dev at lists.digium.com
Sent: Tuesday, October 26, 2004 2:49 PM
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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