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Dinesh Nair

2006-Sep-18 22:53 UTC

[asterisk-users] 488 Not acceptable here sent by Asterisk - SIP debug 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@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@10.14.32.189:5060>
From: <sip:0224580997@10.14.32.179:5060>
;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To: <sip:0224577501@10.14.32.164:5060;user=phone>
Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a@10.14.32.179
CSeq: 1 INVITE
History-Info: <sip:0224577501@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@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@10.14.32.179:5060>
;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To: <sip:0224577501@10.14.32.164:5060;user=phone>;tag=as5a7aa73d
Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a@10.14.32.179
CSeq: 1 INVITE
User-Agent: QubeTalk ECS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0224577501@10.14.32.164>
Content-Length: 0


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


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


-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh@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                                                              |
+=========================================================================+

Steve Langstaff

2006-Sep-19 01:59 UTC

head link

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

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@lists.digium.com 
> [mailto:asterisk-users-bounces@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@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@10.14.32.189:5060>
> From: <sip:0224580997@10.14.32.179:5060> 
> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
> To: <sip:0224577501@10.14.32.164:5060;user=phone>
> Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a@10.14.32.179
> CSeq: 1 INVITE
> History-Info: <sip:0224577501@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@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@10.14.32.179:5060> 
> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
> To: <sip:0224577501@10.14.32.164:5060;user=phone>;tag=as5a7aa73d
> Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a@10.14.32.179
> CSeq: 1 INVITE
> User-Agent: QubeTalk ECS
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:0224577501@10.14.32.164>
> Content-Length: 0
> 
> 
> ---
> Destroying call
'523204-a0e20bd-13c4-132e6-4aed8a6-6d9a@10.14.32.179'
> suria*CLI>
> <-- SIP read from 10.14.32.179:5060:
> ACK sip:0224577501@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@10.14.32.189:5060>
> User-Agent: SIP201 (lp201_sip0423.bin)
> From: <sip:0224580997@10.14.32.179:5060> 
> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
> To: <sip:0224577501@10.14.32.164:5060;user=phone> ;tag=as5a7aa73d
> Call-ID: 523204-a0e20bd-13c4-132e6-4aed8a6-6d9a@10.14.32.179
> CSeq: 1 ACK
> Content-Length:0
> 
> 
> --- (11 headers 0 lines)---
> Destroying call
'523204-a0e20bd-13c4-132e6-4aed8a6-6d9a@10.14.32.179'
>  >>> end sip debug
> 
> 
> -- 
> Regards,                           /\_/\   "All dogs go to
heaven."
> dinesh@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
>

asterisk users - Sep 2006 - 488 Not acceptable here sent by Asterisk - SIP debug follows

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

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