Hi Steve, Yes, that's true. It seems that Asterisk gets it with great delay. For instance:
Asterisk says: ============== Mar 25 11:26:54 VERBOSE[1385] logger.c: Reliably Transmitting (no NAT) to 172.11.11.2:5060: OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0 Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2> Contact: <sip:aster...@172.11.0.201> Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Mar 2010 10:26:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Mar 25 11:26:58 VERBOSE[1385] logger.c: Retransmitting #1 (no NAT) to 172.11.11.2:5060: OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0 Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2> Contact: <sip:aster...@172.11.0.201> Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Mar 2010 10:26:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Mar 25 11:26:58 NOTICE[1385] chan_sip.c: Peer 'MyTestPhone' is now UNREACHABLE! Last qualify: 52 Mar 25 11:26:58 VERBOSE[1385] logger.c: Destroying call '5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201' === WEIRD: Asterisk logs two responses reposnses then (last one at 11:27:02 eventhough last OPTIONS request was sent by Asterisk at 11:26:58): Mar 25 11:26:58 VERBOSE[1385] logger.c: <-- SIP read from 172.11.11.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2>;tag=CD9ADC40-8CECD6C3 CSeq: 102 OPTIONS Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 Contact: <sip:mytestph...@172.11.11.2> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047 Content-Length: 0 Mar 25 11:26:58 VERBOSE[1385] logger.c: --- (10 headers 0 lines) --- Mar 25 11:27:02 VERBOSE[1385] logger.c: <-- SIP read from 172.11.11.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2>;tag=CD9ADC40-8CECD6C3 CSeq: 102 OPTIONS Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 Contact: <sip:mytestph...@172.11.11.2> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047 Content-Length: 0 But Wireshark for the same conversation says: ============================================= >> Sent by Asterisk at 11:26:54.047483000 ----------------------------------------- OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0 Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2> Contact: <sip:aster...@172.11.0.201> Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Mar 2010 10:26:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 << Received from the phone at 11:26:54.097936000 ------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2>;tag=CD9ADC40-8CECD6C3 CSeq: 102 OPTIONS Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 Contact: <sip:mytestph...@172.11.11.2> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047 Content-Length: 0 >> Sent by Asterisk at 11:26:58.486339000 ----------------------------------------- OPTIONS sip:mytestph...@172.11.11.2 SIP/2.0 Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2> Contact: <sip:aster...@172.11.0.201> Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Mar 2010 10:26:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 << Received from the phone at 11:26:58.524907000 ------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport From: "asterisk" <sip:aster...@172.11.0.201>;tag=as5e2d8165 To: <sip:mytestph...@172.11.11.2>;tag=CD9ADC40-8CECD6C3 CSeq: 102 OPTIONS Call-ID: 5bf37c2d2d25a9346cc272d71c3a8...@172.11.0.201 Contact: <sip:mytestph...@172.11.11.2> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047 Content-Length: 0 Hmmmmm, what could be causing such delay between Wireshark getting the data and Asterisk logging it? Regards, Alex -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Thursday, March 25, 2010 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE On 25 Mar 2010, at 10:18, Asterisk wrote: > How is it possible that the peer becames UNREACHABLE eventhough Wireshark > logged its proper response? Wireshark received it, doesn't mean Asterisk did. what does a sip debug in Asterisk show? S -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users