Brian Chrystal
2005-Jan-14 13:12 UTC
[Asterisk-Users] SIP Registration problem, 403 forbidden
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@67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC To: <sip:5622832456@67.110.252.13:5060> CSeq: 1 REGISTER Call-ID: f25ece25-9e450ecf-437df5a2@67.110.253.129 Contact: <sip:5622832456@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@67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC To: <sip:5622832456@67.110.252.13:5060>;tag=as62b71d67 Call-ID: f25ece25-9e450ecf-437df5a2@67.110.253.129 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:5622832456@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@67.110.252.13:5060>' failed for '67.110.253.129' Scheduling destruction of call 'f25ece25-9e450ecf-437df5a2@67.110.253.129' in 15000 ms Destroying call '690879d0-d87de51a-d844291d@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
Brian Chrystal
2005-Jan-14 13:19 UTC
[Asterisk-Users] SIP Registration problem, 403 forbidden
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@67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC To: <sip:5622832456@67.110.252.13:5060> CSeq: 1 REGISTER Call-ID: f25ece25-9e450ecf-437df5a2@67.110.253.129 Contact: <sip:5622832456@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@67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC To: <sip:5622832456@67.110.252.13:5060>;tag=as62b71d67 Call-ID: f25ece25-9e450ecf-437df5a2@67.110.253.129 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:5622832456@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@67.110.252.13:5060>' failed for '67.110.253.129' Scheduling destruction of call 'f25ece25-9e450ecf-437df5a2@67.110.253.129' in 15000 ms Destroying call '690879d0-d87de51a-d844291d@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