[Asterisk-Users] SIP Registration problem, 403 forbidden

[Brian Chrystal]

trying to set up and configure a polycom soundpoint ip 500 phone,  when trying to get it to register with sip, i get the following message


Sip read:
REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" <sip:5622832456 at 67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC
To: <sip:5622832456 at 67.110.252.13:5060>
CSeq: 1 REGISTER
Call-ID: f25ece25-9e450ecf-437df5a2 at 67.110.253.129
Contact: <sip:5622832456 at 67.110.253.129:5060;transport=udp>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0
Max-Forwards: 70
Expires: 300
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 67.110.253.129 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" <sip:5622832456 at 67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC
To: <sip:5622832456 at 67.110.252.13:5060>;tag=as62b71d67
Call-ID: f25ece25-9e450ecf-437df5a2 at 67.110.253.129
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5622832456 at 67.110.252.13>
Content-Length: 0


 to 67.110.253.129:5060
Jan 14 11:44:49 NOTICE[3257]: chan_sip.c:8007 handle_request: Registration from '<sip:5622832456 at 67.110.252.13:5060>' failed for '67.110.253.129'
Scheduling destruction of call 'f25ece25-9e450ecf-437df5a2 at 67.110.253.129' in 15000 ms
Destroying call '690879d0-d87de51a-d844291d at 67.110.253.129'


i'm kinda new to this stuff, so if you need to see any cfg files, let me know and i'll put them up,  thanks