[asterisk-users] 488 Not acceptable here sent by Asterisk - SIP debug follows

[Dinesh Nair]

the situation

Asterisk <-- SIP ---> SIPGW <--- SIP Phone

SIP Phone is trying to call asterisk dialplan:

exten => 0224577501,1,Answer()
exten => 0224577501,2,Playback(demo-instruct)
exten => 0224577501,3,Hangup()

however, asterisk 1.2.12.1 (on FreeBSD 6.1) sends back a "488 Not 
acceptable here" with a CLI message of

WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient information for SDP 
(m = '', c = '')


it seems to be dropping out in process_sdp() because it can't find the m= 
or the c=. this is a little odd, so am wondering if this has triggered some 
edge case in find_sdp(), get_sdp() or get_sdp_iterate(). i've been poring 
thru the code (as the box is remote, and i cant duplicate it locally), but 
can't find exactly where in chan_sip.c its borking.

any advice would be much appreciated.

the SIP debug is attached below:

(10.14.32.179 is the SIPGW, 10.14.32.164 is Asterisk)

 >>> begin sip debug
<-- SIP read from 10.14.32.179:5060:
INVITE sip:0224577501 at 10.14.32.164:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.14.32.179:5060
Via: SIP/2.0/UDP 10.14.32.189:5060
Record-Route: <sip:10.14.32.179:5060>
Supported: replaces
User-Agent: SIP201 (lp201_sip0423.bin)
Contact: <sip:0224580997 at 10.14.32.189:5060>
From: <sip:0224580997 at 10.14.32.179:5060> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To: <sip:0224577501 at 10.14.32.164:5060;user=phone>
Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a at 10.14.32.179
CSeq: 1 INVITE
History-Info: <sip:0224577501 at 10.14.32.164:5060>;index 1
Content-Type: multipart/mixed;boundary=unique-boundary
Content-Length: 474

--unique-boundary
Content-Type: application/sdp

v=0
o=SIP201 12367 0 IN IP4 10.14.32.189
s=SIP201 Session
i=Audio Session
c=IN IP4 10.14.32.189
t=0 0
m=audio 16384 RTP/AVP 4 18 0 8 18
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1

--unique-boundary
Content-Type: application/isup;version=Indonesia
Content-Transfer-Encoding: binary


--- (14 headers 21 lines)---
Using INVITE request as basis request - 
523204-a0e20bd-13c4-132e6-4aed8a6-6d9a at 10.14.32.179
Sending to 10.14.32.179 : 5060 (non-NAT)
Found peer 'RISTI'
Sep 19 09:38:53 WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient 
information for SDP (m = '',
  c = '')
Transmitting (no NAT) to 10.14.32.179:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.14.32.179:5060;received=10.14.32.179
Via: SIP/2.0/UDP 10.14.32.189:5060
From: <sip:0224580997 at 10.14.32.179:5060> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To: <sip:0224577501 at 10.14.32.164:5060;user=phone>;tag=as5a7aa73d
Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a at 10.14.32.179
CSeq: 1 INVITE
User-Agent: QubeTalk ECS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0224577501 at 10.14.32.164>
Content-Length: 0


---
Destroying call '523204-a0e20bd-13c4-132e6-4aed8a6-6d9a at 10.14.32.179'
suria*CLI>
<-- SIP read from 10.14.32.179:5060:
ACK sip:0224577501 at 10.14.32.164:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.14.32.179:5060
Via: SIP/2.0/UDP 10.14.32.189:5060
Record-Route: <sip:10.14.32.179:5060>
Contact: <sip:0224580997 at 10.14.32.189:5060>
User-Agent: SIP201 (lp201_sip0423.bin)
From: <sip:0224580997 at 10.14.32.179:5060> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To: <sip:0224577501 at 10.14.32.164:5060;user=phone> ;tag=as5a7aa73d
Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a at 10.14.32.179
CSeq: 1 ACK
Content-Length:0


--- (11 headers 0 lines)---
Destroying call '523204-a0e20bd-13c4-132e6-4aed8a6-6d9a at 10.14.32.179'
 >>> end sip debug


-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh at alphaque.com                (0 0)   http://www.openmalaysiablog.com/
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