may be the ip phone has the problem, try reset as factory
On Fri, Apr 29, 2011 at 8:03 PM, Mike <l...@net-wall.com> wrote: > 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 >
-- _____________________________________________________________________ -- 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