Hello All, I've spent a day trying to get a Polycom IP500 wokring with my Asterisk box. I have several others that are working fine, but this one is getting by me. Can someone on-list tell from the following SIP debug what I've missed?
Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E From: "2004" <sip:[EMAIL PROTECTED]>;tag=53ED9FBF-D06765E2 To: <sip:[EMAIL PROTECTED];user=phone> CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 225 v=0 o=- 1124335166 1124335166 IN IP4 192.168.1.37 s=Polycom IP Phone c=IN IP4 192.168.1.37 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 14 headers, 10 lines Using latest request as basis request Sending to 192.168.1.37 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.37:2224 Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E From: "2004" <sip:[EMAIL PROTECTED]>;tag=53ED9FBF-D06765E2 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as2c798834 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk", nonce="006c685d" Content-Length: 0 to 192.168.1.37:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '2004' pbx*CLI> Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E From: "2004" <sip:[EMAIL PROTECTED]>;tag=53ED9FBF-D06765E2 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as2c798834 CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines pbx*CLI> Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK893143acACFB06A9 From: "2004" <sip:[EMAIL PROTECTED]>;tag=53ED9FBF-D06765E2 To: <sip:[EMAIL PROTECTED];user=phone> CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="default", realm="asterisk", nonce="006c685d", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="57abe54c660e517d81086bd4f40ad628", algor ithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 225 v=0 o=- 1124335166 1124335166 IN IP4 192.168.1.37 s=Polycom IP Phone c=IN IP4 192.168.1.37 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 15 headers, 10 lines Using latest request as basis request Sending to 192.168.1.37 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.37:2224 Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found user '2004' Aug 17 22:19:30 NOTICE[456]: chan_sip.c:7660 handle_request: Failed to authenticate user "2004" <sip:[EMAIL PROTECTED]>;tag=53ED9FBF-D06765E2 Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK893143acACFB06A9 From: "2004" <sip:[EMAIL PROTECTED]>;tag=53ED9FBF-D06765E2 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as2c798834 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.1.37:5060 pbx*CLI> Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK893143acACFB06A9 From: "2004" <sip:[EMAIL PROTECTED]>;tag=53ED9FBF-D06765E2 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as2c798834 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Thanks, Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
