----- "Asterisk" <aster...@abraxas.si> wrote: > 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
This is interesting as I am seeing the same issue with Snom 360s and M3s on Asterisk 1.6.1.14 and 1.6.2.6. I have also received a report from a colleague who sees a similar issue with Polycoms on Microsoft OCS. -- Thanks, Phil -- _____________________________________________________________________ -- 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