date:20091120

[OpenSIPS-Users] CDRtool freeradius mysql error

2009-11-20 Thread toqeer ali

Hi all,


I successfully configured Freeradius and CDRtool but when i accesses the
CDRs in CDRtool webinterface   i got this error ... it can't get Normalized
field in radacct table of Freeradius...

Please help
 Bellow is the error i got...

"Database error: Invalid SQL: select count(*) as c from radacct where
(AcctStartTime >= '2009-11-21 07:06' and AcctStartTime < '2009-11-21 23:55')
and Normalized = '0' and AcctStopTime != '-00-00 00:00:00' and
(ConnectInfo_stop is not NULL or MediaInfo is NULL or MediaInfo != '' or
(UNIX_TIMESTAMP(NOW()) - UNIX_TIMESTAMP(AcctStopTime) > 20)) MySQL error:
1054 (Unknown column 'Normalized' in 'where clause')"

-- 
Toqeer Ali Syed

Red Hat Certified Engineer
mob: +92 321 9059916
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[OpenSIPS-Users] Another possible bug in b2bua

2009-11-20 Thread Jeff Kronlage

Anca,

 

I'm having the b2b crash at different moments now.  While the system
seems relatively stable if I can get past the initial OK/ACKs, I'm
having what appears to be a problem generated by a to-tag with dashes in
it.  For instance:

 

(IPs/hostnames have been replaced by either (proxy) for our front-end
server or (b2bua) for the back-end b2b or (sip gateway) for our Cisco
gear)

 

15:16:00.217293 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 974) (proxy).5060 > (b2bua).5060: SIP, length: 946

SIP/2.0 200 OK

Via: SIP/2.0/UDP (b2bua);branch=z9hG4bKc4d.78cf4eb3.0

From:
;tag=9e0f6f7cd33d08ad3fd7dcf243da9165-d4dc

To: sip:719xxx1095@(proxy):5060;tag=2E32D030-2CB

Date: Sat, 21 Nov 2009 04:05:21 GMT

Call-ID: B2B.144.0.1258755349

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 2 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

Supported: replaces

Allow-Events: telephone-event

Contact: 

Record-Route:


Content-Type: application/sdp

Content-Length: 244

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 2487 1889 IN IP4 (pstn gateway)

s=SIP Call

c=IN IP4 (pstn gateway)

t=0 0

m=audio 18922 RTP/AVP 0 101

c=IN IP4 (pstn gateway)

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

15:16:00.217949 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 1015) (b2bua).5060 > (proxy).5060: SIP, length: 987

SIP/2.0 200 OK

Record-Route:


Via: SIP/2.0/UDP (proxy);branch=z9hG4bKef3.ffc40151.0

Via: SIP/2.0/UDP
192.168.0.101:5060;rport=52434;received=174.22.143.40;branch=z9hG4bK-c02
4968

From: "719xxx0449"
;tag=a31385f8c7bc6888o0

To: ;tag=B2B.60.0.1258755349

Call-ID: 175b26b8-7cb3...@192.168.0.101

CSeq: 102 INVITE

Content-Type: application/sdp

Supported: replaces

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

Contact: 

Server: OpenSIPS (1.6.1-notls (i386/linux))

Content-Length: 244

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 2487 1889 IN IP4 (pstn gateway)

s=SIP Call

c=IN IP4 (pstn gateway)

t=0 0

m=audio 18922 RTP/AVP 0 101

c=IN IP4 (pstn gateway)

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

15:16:00.299687 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 798) (proxy).5060 > (b2bua).5060: SIP, length: 770

ACK sip:opensips@(b2bua):5060 SIP/2.0

Via: SIP/2.0/UDP (proxy);branch=z9hG4bK-13770017

Via: SIP/2.0/UDP
192.168.0.101:5060;rport=52434;received=174.22.143.40;branch=z9hG4bK-137
70017

From: "719xxx0449"
;tag=a31385f8c7bc6888o0

To: ;tag=B2B.60.0.1258755349

Call-ID: 175b26b8-7cb3...@192.168.0.101

CSeq: 102 ACK

Max-Forwards: 69

User-Agent: Linksys/SPA942-5.1.15(a)

Content-Length: 0

 

** OUTBOUND PACKET GENERATED BY B2B  (note the From: tag, and how it has
a piece of the To: tag stuck in it,repeatedly.) **

15:16:00.299860 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 543) (b2bua).5060 > (proxy).5060: SIP, length: 515

ACK sip:719xxx1095@(pstn gateway):5060 SIP/2.0

Via: SIP/2.0/UDP (b2bua);branch=z9hG4bKc4d.88cf4eb3.0

To: ;tag=2E32D030-2CB

From: <-2CB

From: <-2CB

From: <-2CB

From>;tag=9e0f6f7cd33d08ad3fd7dcf243da9165-d4dc

CSeq: 2 ACK

Call-ID: m:5060>;tag=2E32D030

Route:


Content-Length: 0

User-Agent: OpenSIPS (1.6.1-notls (i386/linux))

Contact: 

 

** The B2BUA Opensips instance crashes here **

 

I honestly don't have a big enough sampling of gear to confirm this.  I
have some softphones, a few handsets, Asterisk, a bunch of Cisco kit and
a Covergence SBC I can talk to.  The softphones and Asterisk work great.
The Cisco kit and Covergence both place dashes in their tags and while
speaking to this gear, the B2B crashes and produces the packet you see
above.

 

Your thoughts?

 

As always, I sure appreciate it.

 

Jeff K

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[OpenSIPS-Users] cdrtool+mediaproxy fromtag-totag mismatch

2009-11-20 Thread erik pepermans

Hi,
 
Installed cdrtool+mediaproxy
 
In the loging I see :
 
Nov 20 23:59:15 ns1 /sbin/opensips[5073]: ACC: transaction answered:
timestamp=1258757955;method=BYE;from_tag=as1638dbb9;to_tag=ccid-879500455-1-
1386;call_id=5288e10b62a2adb7618221930820b...@192.168.5.105;code=200;reason=
OK

In the siptrace I can see :
 
Signalling information  

 Click here to show only
this call id
Call id:5288e10b62a2adb7618221930820b...@192.168.5.105  
  Click here to see the SIP trace for this
call
From/to tags:   as1638dbb9/34eb3f3586e536e54676494def6a9253 
Start time: 2009-11-20 23:58:49 
Stop time:  2009-11-20 23:58:49 
 
The media cannot be seen as from/to tags differ -
 
 
 
 

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believed to be clean.

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Re: [OpenSIPS-Users] Parallel Forking messes up Voicemail two-way audio

2009-11-20 Thread osiris123d


OK.  I ran a trace and see something weird (did a trace also on the loopback
interface and didn't see any traffic at all).  So here is the scenario

The caller is 9XX273XXX8
He is calling 9XX273XXX9 Office Phone (This account has a Mobile number set
up that is 9XX83X3XXX)
The Q-Value for the mobile number is kept in an AVP and the value is 100

So 9XX273XXX9 gets called and the following occurs
lookup("location");  <-- This sets up the 9XX273XXX9 office phone as
the first device to get called.

if ( is_avp_set("$avp(s:mobile_num)") ) 

{   

append_branch("sip:mobile.com");
 
$(branch(uri)[0]) = "sip:" + $avp(s:mobile_num) + "@pt1.vitelity.net";
#Mobile Number URI Config   
$(branch(q)[0]) = $avp(s:mobile_q_value); #Mobile Number Q Value Config 

$(branch(duri)[0]) = "sip:pt1.vitelity.net:5060"; #Mobile Number destination
Config  
};  



serialize_branches(1); <-- Since I only added one number to the
branch (mobile number) it is set up to be called after the office number


failure_route[1] {

if (t_was_cancelled()) {
exit;
};

if (next_branches() )
{
t_on_failure("1");

setbflag(6); # Required to enable Mediaproxy
route(6);
xlog("L_INFO", "Failure_Route 1: In Next Branches Before t_relay ds value is
$ds \n");
t_relay();
}
}


So in the Failure Route you see that the next_branches is getting called. 
So from the SIP trace I performed I believe I see the following

Office Phone 9XX273XXX9 gets called and rings, then after 10 seconds it fail
to the Failure_Route.  The next_branch is true since the mobile number is
the next branch.  So the Mobile number is called, then after 10 seconds it
fails to the Failure_Route.   ( Here is the weird thing I notice )
The next_branch is true again and the office number is the value!?  How did
the office number get added to the branch?  If(next_branches()) should have
been false.

Here is the SIP Trace (I also included the OpenSIPS.log stuff from my xlog
messages)

Main Route: call CallId [ZGVjZmYzMWU3YjBjNGJmMzY5ZTkwZDAwNThmZWFjY2U.]
[INVITE] rU[273XXX9] ru[sip:273x...@coolbeans.com]
fu[sip:9xx273x...@coolbeans.com] tu[sip:273x...@coolbeans.com]
si[192.XXX.XXX.XXX] ct[]
RPID[] 
Route 3: Call [INVITE] rU[273XXX9] ru[sip:273x...@coolbeans.com]
fu[sip:9xx273x...@coolbeans.com] tu[sip:273x...@coolbeans.com]
si[192.XXX.XXX.XXX] ct[] 

U 192.XXX.XXX.XXX:13139 -> 6X.XX.XXX.XX:5060
INVITE sip:273x...@coolbeans.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.82:63250;branch=z9hG4bK-d8754z-3c9ca166894e6fcf-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: .
From: "Bob Yoda";tag=9d9dbc27.
Call-ID: ZGVjZmYzMWU3YjBjNGJmMzY5ZTkwZDAwNThmZWFjY2U..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
User-Agent: Bria release 2.5.4 stamp 53956.
Content-Length: 260.
.
v=0.
o=- 4 2 IN IP4 192.168.100.82.
s=CounterPath Bria.
c=IN IP4 192.168.100.82.
t=0 0.
m=audio 34314 RTP/AVP 107 0 8 18 101.
a=sendrecv.
a=rtpmap:107 BV32/16000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 6X.XX.XXX.XX:5060 -> 192.XXX.XXX.XXX:13139
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.100.82:63250;branch=z9hG4bK-d8754z-3c9ca166894e6fcf-1---d8754z-;rport=13139;received=192.XXX.XXX.XXX.
To: ;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.a3ec.
From: "Bob Yoda";tag=9d9dbc27.
Call-ID: ZGVjZmYzMWU3YjBjNGJmMzY5ZTkwZDAwNThmZWFjY2U..
CSeq: 1 INVITE.
Proxy-Authenticate: Digest realm="coolbeans.com",
nonce="4b05b71a0004eae52f6f8bc6f96c164aef2a6b20b5d4", qop="auth".
Server: Ae SIP Proxy.
Content-Length: 0.
.


U 192.XXX.XXX.XXX:13139 -> 6X.XX.XXX.XX:5060
ACK sip:273x...@coolbeans.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.82:63250;branch=z9hG4bK-d8754z-3c9ca166894e6fcf-1---d8754z-;rport.
Max-Forwards: 70.
To: ;tag=eb4f595d3de5f0bd1aa7f6687dc18bd8.a3ec.
From: "Bob Yoda";tag=9d9dbc27.
Call-ID: ZGVjZmYzMWU3YjBjNGJmMzY5ZTkwZDAwNThmZWFjY2U..
CSeq: 1 ACK.
Content-Length: 0.
.












Main Route: call CallId [ZGVjZmYzMWU3YjBjNGJmMzY5ZTkwZDAwNThmZWFjY2U.]
[INVITE] rU[273XXX9] ru[sip:273x...@coolbeans.com]
fu[sip:9xx273x...@coolbeans.com] tu[sip:273x...@coolbeans.com]
si[192.XXX.XXX.XXX] ct[]
RPID[] 
Route 3: Call [INVITE] rU[273XXX9] ru[sip:273x...@coolbeans.com]
fu[sip:9xx273x...@coolbeans.com] tu[sip:273x...@coolbeans.com]
si[192.XXX.XXX.XXX] ct[] 
Route 10: Call [INVITE] rU[901273XXX9] ru[sip:901273x...@irock.com]
rd[irock.com] fu[sip:9xx273x...@coolbeans.com] tu[sip:273x...@coolbeans.com]
si[192.XXX.XXX.XXX] ct[]
RPID[]

[OpenSIPS-Users] Radius repeats the record to CDRTool/Radius

2009-11-20 Thread Alberto Listas

Hi All,

When I make a call opensips sends the radius info to freeradius without any 
indication of error.
But a strange thing happens, when I look at the radius.log in the freeradius 
server
I get between 2 and 100 times these repeated messages:

Fri Nov 20 18:14:31 2009 : Error: rlm_sql_mysql: MySQL error 'Duplicate entry 
'NWFiYTY4NWVmYWZlNjc4OTYyNWMxZWMwZWZlZjIwYWI.-7f47744f-as27abee74' for key$
Fri Nov 20 18:14:31 2009 : Error: rlm_sql (sql): Couldn't insert SQL accounting 
START record - Duplicate entry 'NWFiYTY4NWVmYWZlNjc4OTYyNWMxZWMwZWZlZjIw$
Fri Nov 20 18:14:31 2009 : Error: rlm_sql_mysql: Cannot store result

And this increments the KBIn and KBOut in CDRTool
It looks like the radiusclient-ng is sending the same record sometimes more 
than 100 times !!! 

Does anyone have a suggestion?

Thanks,

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Re: [OpenSIPS-Users] CDRTool/Radius Problem

2009-11-20 Thread Alberto Listas

Hi Laszlo,

It was a problem with the dictionaries! Thanks for the suggestion.
I also fixed an error in the config for the new aaa_radius, and now
I get the records in the Radius/CDRTool . But a strange thing happens
when I look at the radius.log I get between 3 and 100 times these repeated
messages:

Fri Nov 20 18:14:31 2009 : Error: rlm_sql_mysql: MySQL error 'Duplicate entry 
'NWFiYTY4NWVmYWZlNjc4OTYyNWMxZWMwZWZlZjIwYWI.-7f47744f-as27abee74' for key$
Fri Nov 20 18:14:31 2009 : Error: rlm_sql (sql): Couldn't insert SQL accounting 
START record - Duplicate entry 'NWFiYTY4NWVmYWZlNjc4OTYyNWMxZWMwZWZlZjIw$
Fri Nov 20 18:14:31 2009 : Error: rlm_sql_mysql: Cannot store result

And this increments the KBIn and KBOut in CDRTool
It looks like the radiusclient-ng is sending the same record more
than 100 times !!! I am going to post this as another issue.

Thanks,

Alberto


  - Original Message - 
  From: Laszlo 
  To: OpenSIPS users mailling list 
  Sent: Thursday, November 19, 2009 7:19 PM
  Subject: Re: [OpenSIPS-Users] CDRTool/Radius Problem


  Hi Alberto,


  2009/11/19 Alberto Listas 

Hi,

I am installing Opensips v 1.6.0, MediaProxy v 2.3.9 and CDRTool v 6.9.9 on 
a test system.
When a call finishes I get an error in the Opensips log:

Nov 19 20:37:58 os4 media-dispatcher[9116]: error: failed to send radius 
accounting record: 'Sip-From-Tag'

And Nothing gets added to the CDRTool or Radius Database. In the old 
versions I got error
from /sbin/opensips when radius had trouble not from media-dispatcher. I 
know Radius
is working  because I can telnet in to the radius server. I get no errors 
in the CDRTool/Radius
Server log.

Does anyone have a suggestion?

Thanks,

Alberto



  Looks like a problem with the radius dictionaries.


  -Laszlo



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[OpenSIPS-Users] OpenSIPs CP MDB2 and xmlrpc Errors

2009-11-20 Thread Sajjad Ali Mushtaq

Hello
i am novice opensips user
and 
I am using OpenSIPs 1.6.0 with OpenSIPs-CP

In  Installation went fine but when i edit an entry in
load balancer module it gives an error "" Failed to issue query, error
message: MDB2 error: syntax error


In the OpenSIPs.cfg file 
xmlrpc module is loaded  as 
""
loadmodule "mi_xmlrpc.so"  
modparam("mi_xmlrpc", "log_file", "/var/log/abyss.log")
modparam("mi_xmlrpc", "reply_option", 0)
modparam("mi_xmlrpc", "buffer_size", 8192)

""
and i have checked the syslog file and there is some info about mi_xmlrpc 
module as

"""  Nov 20 18:17:58 ali-ub-910-1-e-desktop ./sbin/opensips[2703]: 
INFO:mi_xmlrpc:xmlrpc_process: starting xmlrpc server  """

but when i run the command
netstat -tal

there is nothing running on port 8080


Additionally when i access the Domain Link the error then is 

"" Fatal error: Call to undefined method MDB2_Error::setFetchMode() in 
/var/www/opensips-cp/web/tools/domains/lib/db_connect.php on line 16 ""


and the lin 16 in the file is ""  $link->setFetchMode(MDB2_FETCHMODE_ASSOC);  ""


when i try to access statistics monitor the error
then is ""  Failed to issue query, error message : MDB2 Error: no such
table ""


please guide me to resolve the problem


thanks in advance for your help


Best Regards



  

Sajjad Ali Mushtaq

email:  alisajjad_mush...@yahoo.com

Mobile: 0033667961001


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Re: [OpenSIPS-Users] Avoiding use of table presentity

2009-11-20 Thread mayamatakeshi

Thanks, Anka,
I got it.

best regards.

On Fri, Nov 20, 2009 at 10:14 PM, Anca Vamanu  wrote:

> mayamatakeshi wrote:
> > Hello,
> > I'm experimenting with module presence.
> > Is it possible somehow to make presence to NOT write to table presentity?
> > I'm sending PUBLISH messages to opensips and I just want its contents
> > to be sent to the watchers. I don't care to add previous contents nor
> > even to send anything in case of NOTIFY due to SUBSCRIBE.
> >
> > regards,
> > takeshi
> > 
> >
> > ___
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> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> Hi Takeshi,
>
> No, it is not possible to avoid using presentity table. The presence
> module uses it real time for storing info from Publish messages and
> getting info when sending Notifies. The reason why the database must be
> used in this case is that the documents can be large and can not be
> stored in memory.
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
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Re: [OpenSIPS-Users] Any word on the B2B bug?

2009-11-20 Thread Jeff Kronlage

Anca,

Thanks *so* much - I really appreciate your effort!

--
Jeff Kronlage
Data102


-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Friday, November 20, 2009 9:07 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Any word on the B2B bug?

Hi Jeff,

I have fixed it (with a bit of delay than what I promised :) ). Please 
update from svn.

Thanks and regards,
Anca

Jeff Kronlage wrote:
>
> Hi Anca,
>
> One of my coworkers is on the devel mailing list and saw the repeated 
> "OK" issue I was having with the B2B come through as a bug report. 
> Don't mean to be a pest or to rush anyone, but we were curious if 
> there was a timeframe on that being addressed? It would help to know 
> for our internal planning.
>
> Thanks so much,
>
> Jeff K
>
>

>
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-- 
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Re: [OpenSIPS-Users] Any word on the B2B bug?

2009-11-20 Thread Anca Vamanu

Hi Jeff,

I have fixed it (with a bit of delay than what I promised :) ). Please 
update from svn.

Thanks and regards,
Anca

Jeff Kronlage wrote:
>
> Hi Anca,
>
> One of my coworkers is on the devel mailing list and saw the repeated 
> “OK” issue I was having with the B2B come through as a bug report. 
> Don’t mean to be a pest or to rush anyone, but we were curious if 
> there was a timeframe on that being addressed? It would help to know 
> for our internal planning.
>
> Thanks so much,
>
> Jeff K
>
> 
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
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www.voice-system.ro


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[OpenSIPS-Users] dialog - failed to get From header

2009-11-20 Thread Jeff Pyle

Hello,

I'm seeing messages like this from time to time on Opensips 1.5.3:

  ERROR:dialog:pre_match_parse: failed to get From header

My dialog config is as follows:

modparam("dialog", "dlg_flag", 4)
modparam("dialog", "dlg_match_mode", 2)
modparam("dialog", "db_url", [functioning db_uri])
modparam("dialog", "db_mode", 1)
modparam("dialog", "profiles_with_value", "inbound; outbound")

This proxy took over for an old Openser 1.3.2 box who didn't complain about
any From header problems.  I can't imagine I'm receiving INVITEs without
>From headers.  Unfortunately it's too sporadic to enable full debugging.
Any thoughts?


Thanks,
Jeff


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Re: [OpenSIPS-Users] Avoiding use of table presentity

2009-11-20 Thread Anca Vamanu

mayamatakeshi wrote:
> Hello,
> I'm experimenting with module presence.
> Is it possible somehow to make presence to NOT write to table presentity?
> I'm sending PUBLISH messages to opensips and I just want its contents 
> to be sent to the watchers. I don't care to add previous contents nor 
> even to send anything in case of NOTIFY due to SUBSCRIBE.
>
> regards,
> takeshi
> 
>
> ___
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>   
Hi Takeshi,

No, it is not possible to avoid using presentity table. The presence 
module uses it real time for storing info from Publish messages and 
getting info when sending Notifies. The reason why the database must be 
used in this case is that the documents can be large and can not be 
stored in memory.

Regards,

-- 
Anca Vamanu
www.voice-system.ro

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[OpenSIPS-Users] t_check_trans and CANCEL

2009-11-20 Thread Lasse Johnsen

Hi,

I'm having a hard time getting t_check_trans to recognise a particular 
transaction.

I have a scenario where the client is behind NAT. In this instance, I cannot 
find a way to properly relay CANCEL.

In my main routing block I have:

 if (is_method("CANCEL")){
if ( t_check_trans() ) {
t_relay();
}
}

The problem is that my device uses it's internal RFC1918 IP in the Via in the 
CANCEL.

The invite uses the external IP in the Via.

INVITE: 

Session Initiation Protocol
Request-Line: INVITE sip:447832335...@193.104.93.31 SIP/2.0
Method: INVITE
Request-URI: sip:447832335...@193.104.93.31
Request-URI User Part: 447832335392
Request-URI Host Part: 193.104.93.31
[Resent Packet: False]
Message Header
Max-Forwards: 70
From: ;tag=88A1D89809DC70828B238BD874369F6C
SIP from address: sip:la...@193.104.93.31
SIP from address User Part: lasse
SIP from address Host Part: 193.104.93.31
SIP tag: 88A1D89809DC70828B238BD874369F6C
Contact: 
Contact Binding: 
URI: 
SIP contact address: sip:la...@87.194.36.133:49477
Allow: ACK
Allow: CANCEL
Allow: BYE
User-Agent: Softphone version 2.6
To: 
SIP to address: sip:447832335...@193.104.93.31
SIP to address User Part: 447832335392
SIP to address Host Part: 193.104.93.31
Content-Type: application/sdp
Call-ID: 2803C4C57FC42DD6C26F450AD4F2558016AA63CA
CSeq: 1 INVITE
Sequence Number: 1
Method: INVITE
Via: SIP/2.0/UDP 87.194.36.133:49477;branch=z9hG4bK2;rport
Transport: UDP
Sent-by Address: 87.194.36.133
Sent-by port: 49477
Branch: z9hG4bK2
RPort: rport
Content-Length: 182

The cancel has an internal RFC1918 IP address:

CANCEL:

Session Initiation Protocol
Request-Line: CANCEL sip:447832335...@193.104.93.31 SIP/2.0
Method: CANCEL
Request-URI: sip:447832335...@193.104.93.31
Request-URI User Part: 447832335392
Request-URI Host Part: 193.104.93.31
[Resent Packet: False]
Message Header
Call-ID: 2803C4C57FC42DD6C26F450AD4F2558016AA63CA
To: 
SIP to address: sip:447832335...@193.104.93.31
SIP to address User Part: 447832335392
SIP to address Host Part: 193.104.93.31
From: ;tag=88A1D89809DC70828B238BD874369F6C
SIP from address: sip:la...@193.104.93.31
SIP from address User Part: lasse
SIP from address Host Part: 193.104.93.31
SIP tag: 88A1D89809DC70828B238BD874369F6C
CSeq: 1 CANCEL
Sequence Number: 1
Method: CANCEL
Via: SIP/2.0/UDP 192.168.0.200:49477;branch=z9hG4bK2;rport
Transport: UDP
Sent-by Address: 192.168.0.200
Sent-by port: 49477
Branch: z9hG4bK2
RPort: rport
Max-Forwards: 70
Content-Length: 0


I have also tried the following:

 if (is_method("CANCEL")){
force_rport();
add_rcv_param();
fix_nated_contact();
if ( t_check_trans() ) {
t_relay();
}
}

This doesn't make a difference.

The RFC3261branch in both INVITE and CANCEL magic cookie is identical 
(z9hG4bK2).

in modules/tm/t_lookup.c we have:

static inline int via_matching( struct via_body *inv_via,
struct via_body *ack_via )
..
if (memcmp(inv_via->host.s, ack_via->host.s,
ack_via->host.len)!=0)
return 0;

When our CANCEL passes then: 

ack_via->host.s is 192.168.0.200
inv_via->host.s is 87.194.36.133

Therefore there is no match. 

In the instance where we remove this host check the CANCEL is processed 
correctly.

How can this be resolved more reasonably?


Sincerely,


Lasse











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[OpenSIPS-Users] Avoiding use of table presentity

2009-11-20 Thread mayamatakeshi

Hello,
I'm experimenting with module presence.
Is it possible somehow to make presence to NOT write to table presentity?
I'm sending PUBLISH messages to opensips and I just want its contents to be
sent to the watchers. I don't care to add previous contents nor even to send
anything in case of NOTIFY due to SUBSCRIBE.

regards,
takeshi
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