[asterisk-users] 488 Not acceptable here sent by Asterisk -
SIPdebug follows
Steve Langstaff
steve.langstaff at citel.com
Tue Sep 19 01:59:06 MST 2006
I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124&nbn=4
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Dinesh Nair
> Sent: 19 September 2006 06:54
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] 488 Not acceptable here sent by
> Asterisk - SIPdebug follows
>
>
> 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/
> +==========================----oOO--(_)--OOo----==============
> ============+
> | for a in past present future; do
> |
> | for b in clients employers associates relatives
> neighbours pets; do |
> | echo "The opinions here in no way reflect the opinions of
> my $a $b." |
> | done; done
> |
> +=============================================================
> ============+
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list