Just a follow-up in case somebody else sees this: I upgraded the Polycom phone to the latest firmware, that did it. I had been on the same version for almost a year without problems, so I don`t know if it`s the firmware version that was the issue or simply formatting the phone to factory default would have fixed it .
Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, April 29, 2011 11:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP bad request What I am looking for? Here is a snippet, with some info obfuscated. I can see the bad request, but why there is such a message isn’t obvious. <--- SIP read from UDP:23.23.23.23:23725 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: "JOHN SMITH" <sip:5555555555@66.66.66.66>;tag=as40e0c5af To: "user4444" <sip:user4444@192.168.1.90:5060>;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: <sip:user4444@192.168.1.90:5060> User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Allow-Events: talk,hold,conference Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:23.23.23.23:23725 ---> SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: "JOHN SMITH" <sip:5555555555@66.66.66.66>;tag=as40e0c5af To: "user4444" <sip:user4444@192.168.1.90:5060>;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: <sip:user4444@192.168.1.90:5060> User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ??????? ????? Sent: Friday, April 29, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP bad request Try to look in 'sip set debug peer user4444'. On 29.04.2011 18:10, Mike wrote: Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user4444-0006fcdd is ringing -- Got SIP response 400 "Bad Request" back from 23.23.23.23 -- SIP/user4444-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of “bad request”? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _____________________________________________________________________ -- 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