[asterisk-users] SIP bad request

Wed May 4 12:16:59 CDT 2011

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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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 at 66.66.66.66>;tag=as40e0c5af

To: "user4444" <sip:user4444 at 192.168.1.90:5060>;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863 at 66.66.66.66

Contact: <sip:user4444 at 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 at 66.66.66.66>;tag=as40e0c5af

To: "user4444" <sip:user4444 at 192.168.1.90:5060>;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863 at 66.66.66.66

Contact: <sip:user4444 at 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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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

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