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Summary:ASTERISK-09047: 'Got SIP response 400 "Bad Request" back from 192.168.1.206' where calling a Thomson 2030
Reporter:Iñaki Baz Castillo (ibc)Labels:
Date Opened:2007-03-19 12:48:53Date Closed:2011-06-07 14:07:26
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:In my network I've configured lots of SJphone, Twinkle and Ekiga softphones, a Linksys PAP2 and a Linksys SPA942 working fine and making calls between them with Asterisk.

Now I've installed a Thomson 2030 (and upgraded to firmware 1.52). It can call properly and receive calls from Asterisk except when it receives calls from any SJphone or Linksys SPA942. Then the following occurs:

asterisk2*CLI>
   -- Executing [102@desde-usuarios:1] Dial("SIP/sjphone-softphone-0820a1f8", "SIP/thomson2030-tfno-ip|600|wtWT") in new stack
   -- Called thomson2030-tfno-ip
   -- Got SIP response 400 "Bad Request" back from 192.168.1.206
   -- SIP/thomson2030-tfno-ip-082050b8 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
 == Auto fallthrough, channel 'SIP/sjphone-softphone-0820a1f8' status is 'CONGESTION'

Note that it work fine if I call from the Linksys PAP2, Twinkle or Ekiga to the Thomson.

I attach the complete SIP debug in "Additional Information".

****** ADDITIONAL INFORMATION ******

DEBUG INFO:
***********


CLI> sip show peers
 sjphone-softphone/sjphone-s  192.168.1.63     D          5060     OK (1 ms)
 thomson2030-tfno-ip/thomson2030-t  192.168.1.206    D          5060     OK (8 ms)



************************************************************************************************************************
 -- Executing [102@desde-usuarios:1] Dial("SIP/sjphone-softphone-0820a1f8", "SIP/thomson2030-tfno-ip|600|wtWT") in new stack
************************************************************************************************************************


Audio is at 192.168.1.203 port 11518
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.206:5060:

########################################################################################
INVITE sip:thomson2030-tfno-ip@192.168.1.206:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK0428db07;rport
From: "Adrian <116>" <sip:sjphone-softphone@192.168.1.203>;tag=as1d5e95ee
To: <sip:thomson2030-tfno-ip@192.168.1.206:5060;user=phone>
Contact: <sip:sjphone-softphone@192.168.1.203>
Call-ID: 210877495311a2dc45a0530c2eba087c@192.168.1.203
CSeq: 102 INVITE
User-Agent: NEW Ilimit - Asterisk PBX
Max-Forwards: 70
Date: Mon, 19 Mar 2007 16:36:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 14934 14934 IN IP4 192.168.1.203
s=session
c=IN IP4 192.168.1.203
t=0 0
m=audio 11518 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
########################################################################################

************************************************************************************************************************
   -- Called thomson2030-tfno-ip
************************************************************************************************************************

<--- SIP read from 192.168.1.206:5060 --->

########################################################################################
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK0428db07;rport
From: "Adrian"<sip:>
To: <sip:thomson2030-tfno-ip@192.168.1.206:5060;user=phone>
Call-ID: 210877495311a2dc45a0530c2eba087c@192.168.1.203
CSeq: 102 INVITE
Content-Length: 0
########################################################################################

<------------->
--- (7 headers 0 lines) ---

************************************************************************************************************************
   -- Got SIP response 400 "Bad Request" back from 192.168.1.206
************************************************************************************************************************

Transmitting (no NAT) to 192.168.1.206:5060:

########################################################################################
ACK sip:thomson2030-tfno-ip@192.168.1.206:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK0428db07;rport
From: "Adrian <116>" <sip:sjphone-softphone@192.168.1.203>;tag=as1d5e95ee
To: <sip:thomson2030-tfno-ip@192.168.1.206:5060;user=phone>
Contact: <sip:sjphone-softphone@192.168.1.203>
Call-ID: 210877495311a2dc45a0530c2eba087c@192.168.1.203
CSeq: 102 ACK
User-Agent: NEW Ilimit - Asterisk PBX
Max-Forwards: 70
Content-Length: 0
########################################################################################

---

************************************************************************************************************************
   -- SIP/thomson2030-tfno-ip-082100a8 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
 == Auto fallthrough, channel 'SIP/sjphone-softphone-0820a1f8' status is 'CONGESTION'
************************************************************************************************************************

Really destroying SIP dialog '210877495311a2dc45a0530c2eba087c@192.168.1.203' Method: INVITE
Really destroying SIP dialog '075EC4F3-D284-4A64-B8A5-0CD40256299E@192.168.1.63' Method: ACK
<--- SIP read from 192.168.1.206:5060 --->

########################################################################################
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK0428db07;rport
From: "Adrian"<sip:>
To: <sip:thomson2030-tfno-ip@192.168.1.206:5060;user=phone>
Call-ID: 210877495311a2dc45a0530c2eba087c@192.168.1.203
CSeq: 102 ACK
Content-Length: 0
########################################################################################

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '4b0c6a6a1a57d4410f7ca3087f0d9ba9@192.168.1.203' Method: OPTIONS
asterisk2*CLI>

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Comments:By: Serge Vecher (serge-v) 2007-03-19 13:07:31

Well, the error message is "Bad Request" coming back from the Thomson. That's where the problem lies. Please contact the phone manufacturer for support.