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

2006-09-22 Thread Dinesh Nair 




On 09/20/06 15:06 Dinesh Nair said the following:



On 09/19/06 16:59 Steve Langstaff said the following:


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=7124nbn=4



thanks for the link,

however, on 18th may 2006, kpfleming's note says, This should be fixed 
in both 1.2 branch and trunk, and i'm using 1.2.12.1 which was just 
released this week. looking thru the current chan_sip.c code, it does 
seem like kevin's modified patch has been committed into the branch i'm 
using, so this isnt the problem.


[am cc'ing reply into -dev because a bug report was opened on this at 
http://bugs.digium.com/view.php?id=8010 with a patch provided]


i've managed to track this down to a loop which terminated prematurely in 
find_sdp() in chan_sip.c. this bug would have prevented proper handling of 
multipart/mixed content types due to the loop which searches for the end of 
the block ending prematurely and setting req-sdp_start  req-sdp_end.


i've provided patches for trunk and 1.2.x in the bug entry, as i think this 
should also be committed to 1.2.x.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](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

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

2006-09-20 Thread Dinesh Nair 



On 09/19/06 16:59 Steve Langstaff said the following:

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=7124nbn=4


thanks for the link,

however, on 18th may 2006, kpfleming's note says, This should be fixed in 
both 1.2 branch and trunk, and i'm using 1.2.12.1 which was just released 
this week. looking thru the current chan_sip.c code, it does seem like 
kevin's modified patch has been committed into the branch i'm using, so 
this isnt the problem.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](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

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

2006-09-19 Thread Steve Langstaff 
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=7124nbn=4


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060
 From: sip:[EMAIL PROTECTED]:5060 
 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
 To: sip:[EMAIL PROTECTED]:5060;user=phone
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 INVITE
 History-Info: sip:[EMAIL PROTECTED]: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 - 
 [EMAIL PROTECTED]
 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:[EMAIL PROTECTED]:5060 
 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as5a7aa73d
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 INVITE
 User-Agent: QubeTalk ECS
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Contact: sip:[EMAIL PROTECTED]
 Content-Length: 0
 
 
 ---
 Destroying call '[EMAIL PROTECTED]'
 suria*CLI
 -- SIP read from 10.14.32.179:5060:
 ACK sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060
 User-Agent: SIP201 (lp201_sip0423.bin)
 From: sip:[EMAIL PROTECTED]:5060 
 ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
 To: sip:[EMAIL PROTECTED]:5060;user=phone ;tag=as5a7aa73d
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 ACK
 Content-Length:0
 
 
 --- (11 headers 0 lines)---
 Destroying call '[EMAIL PROTECTED]'
   end sip debug
 
 
 -- 
 Regards,   /\_/\   All dogs go to heaven.
 [EMAIL PROTECTED](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
 
___
--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