Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-30 Thread Ahmed Munir
Hi.

Thanks for your reply, the traces are metioned below;

U 203.215.176.22:55134 - 11.22.33.44:5060
.
.
..

U 81.201.82.45:5060 - 11.22.33.44:5060
INVITE sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 SIP/2.0.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 INVITE.
From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
;tag=43772.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44.
Via: SIP/2.0/UDP 81.201.82.45:5060
;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
Max-Forwards: 69.
Content-Type: application/sdp.
Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
User-Agent: Vox Callcontrol.
Content-Length: 210.
.
v=0.
o=root 13293 13293 IN IP4 81.201.82.146.
s=session.
c=IN IP4 81.201.82.146.
t=0 0.
m=audio 11458 RTP/AVP 8 0.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


U 11.22.33.44:5060 - 81.201.82.45:5060
SIP/2.0 100 Giving a try.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 INVITE.
From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
;tag=43772.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44.
Via: SIP/2.0/UDP 81.201.82.45:5060
;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
Server: OpenSIPS (1.6.1-notls (i386/linux)).
Content-Length: 0.
.


U 11.22.33.44:5060 - 203.215.176.22:55134
INVITE sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
Record-Route: sip:11.22.33.44;lr=on.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 INVITE.
From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
;tag=43772.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
Via: SIP/2.0/UDP 81.201.82.45:5060
;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
Max-Forwards: 68.
Content-Type: application/sdp.
Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
User-Agent: Vox Callcontrol.
Content-Length: 210.
P-hint: usrloc applied.
.
v=0.
o=root 13293 13293 IN IP4 81.201.82.146.
s=session.
c=IN IP4 81.201.82.146.
t=0 0.
m=audio 11458 RTP/AVP 8 0.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


U 203.215.176.22:55134 - 11.22.33.44:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
Via: SIP/2.0/UDP 81.201.82.45:5060
;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
Record-Route: sip:11.22.33.44;lr.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
From: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 INVITE.
User-Agent: X-Lite release 1104o stamp 56125.
Content-Length: 0.
.


U 11.22.33.44:5060 - 81.201.82.45:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 81.201.82.45:5060
;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
Record-Route: sip:11.22.33.44;lr.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
From: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 INVITE.
User-Agent: X-Lite release 1104o stamp 56125.
Content-Length: 0.
.


U 203.215.176.22:55134 - 11.22.33.44:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
Via: SIP/2.0/UDP 81.201.82.45:5060
;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
Record-Route: sip:11.22.33.44;lr.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
From: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
User-Agent: X-Lite release 1104o stamp 56125.
Content-Length: 130.
.
v=0.
o=- 2 2 IN IP4 192.168.0.168.
s=CounterPath X-Lite 3.0.
c=IN IP4 192.168.0.168.
t=0 0.
m=audio 1876 RTP/AVP 8 0.
a=sendrecv.


U 11.22.33.44:5060 - 81.201.82.45:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 81.201.82.45:5060
;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
Record-Route: sip:11.22.33.44;lr.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
From: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
User-Agent: X-Lite release 1104o stamp 56125.

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-30 Thread Laszlo
Hi Ahmed,

As you can see, the other party gets local ip in SDP

c=IN IP4 192.168.0.168.

You can try to play with flags:
http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028

-Laszlo



2010/4/30 Ahmed Munir ahmedmunir...@gmail.com



 Hi.

 Thanks for your reply, the traces are metioned below;

 U 203.215.176.22:55134 - 11.22.33.44:5060
 .
 .
 ..

 U 81.201.82.45:5060 - 11.22.33.44:5060
 INVITE sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 SIP/2.0.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
 Max-Forwards: 69.
 Content-Type: application/sdp.
 Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
 User-Agent: Vox Callcontrol.
 Content-Length: 210.
 .
 v=0.
 o=root 13293 13293 IN IP4 81.201.82.146.
 s=session.
 c=IN IP4 81.201.82.146.
 t=0 0.
 m=audio 11458 RTP/AVP 8 0.
 a=rtpmap:8 PCMA/8000.
 a=rtpmap:0 PCMU/8000.
 a=silenceSupp:off - - - -.
 a=ptime:20.
 a=sendrecv.


 U 11.22.33.44:5060 - 81.201.82.45:5060
 SIP/2.0 100 Giving a try.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
 Server: OpenSIPS (1.6.1-notls (i386/linux)).
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 203.215.176.22:55134
 INVITE sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
 Record-Route: sip:11.22.33.44;lr=on.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
 Max-Forwards: 68.
 Content-Type: application/sdp.
 Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
 User-Agent: Vox Callcontrol.
 Content-Length: 210.
 P-hint: usrloc applied.
 .
 v=0.
 o=root 13293 13293 IN IP4 81.201.82.146.
 s=session.
 c=IN IP4 81.201.82.146.
 t=0 0.
 m=audio 11458 RTP/AVP 8 0.
 a=rtpmap:8 PCMA/8000.
 a=rtpmap:0 PCMU/8000.
 a=silenceSupp:off - - - -.
 a=ptime:20.
 a=sendrecv.


 U 203.215.176.22:55134 - 11.22.33.44:5060
 SIP/2.0 180 Ringing.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
 Record-Route: sip:11.22.33.44;lr.
 Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 User-Agent: X-Lite release 1104o stamp 56125.
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 81.201.82.45:5060
 SIP/2.0 180 Ringing.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
 Record-Route: sip:11.22.33.44;lr.
 Contact: sip:4...@203.215.176.22:55134
 ;rinstance=25bfe05618433c26;nat=yes.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 User-Agent: X-Lite release 1104o stamp 56125.
 Content-Length: 0.
 .


 U 203.215.176.22:55134 - 11.22.33.44:5060
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
 Record-Route: sip:11.22.33.44;lr.
 Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO.
 Content-Type: application/sdp.
 User-Agent: X-Lite release 1104o stamp 56125.
 Content-Length: 130.
 .
 v=0.
 o=- 2 2 IN IP4 192.168.0.168.
 s=CounterPath X-Lite 3.0.
 c=IN IP4 192.168.0.168.
 t=0 0.
 m=audio 1876 RTP/AVP 8 0.
 a=sendrecv.


 U 11.22.33.44:5060 - 81.201.82.45:5060
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
 Record-Route: sip:11.22.33.44;lr.
 Contact: sip:4...@203.215.176.22:55134
 ;rinstance=25bfe05618433c26;nat=yes.

Re: [OpenSIPS-Users] spam on callcontrol website

2010-04-30 Thread Adrian Georgescu
Fixed thanks

Adrian

On Apr 30, 2010, at 4:30 AM, Paul Wise wrote:

 Hi all,

 I found some more spam, this time on the callcontrol website. The  
 front
 page (one essay link), Installation page (two essay links) and the
 tickets (three german spams).

 -- 
 bye,
 pabs

 http://bonedaddy.net/pabs3/
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Re: [OpenSIPS-Users] cdrtool rating computation problem

2010-04-30 Thread Adrian Georgescu
You should use call control by using internally 00+E164 format

Adrian


On Apr 30, 2010, at 6:12 AM, Paul Wise wrote:

 I'm having this problem:

 http://opensips.wordpress.com/2009/06/26/cdrtool-rating-computation-problem/

 Looking at the code, it seems to be caused by rating.php line 6443:

if (!preg_match(/^0/,$CDR-CanonicalURINormalized)) {

 Should I be working around this line of code by prefixing a 0 to the
 can_uri avp before calling call_control() in opensips.cfg?

 -- 
 bye,
 pabs

 http://bonedaddy.net/pabs3/
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Re: [OpenSIPS-Users] Autehntification from DB with AVPOPS module ?

2010-04-30 Thread samoh

Hi Bogdan,

Thank you for your response.

Sam.
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Re: [OpenSIPS-Users] CDRTool Rating: Does not rate some calls

2010-04-30 Thread Mike O'Connor
On 28/04/10 6:56 PM, Mike O'Connor wrote:
 Hi Adrian
   
 Where as a call to 132221 (ie the special) is rated as '
 diverted-on-net' and is displayed as
 132...@sip.xxx.xx.xx
   
 This means that in your OpenSIPS configuration login did not set
 correctly the CanonicalURI.

 
 Ok taking your explaination above I edited the CanonicalURI to say
 0061132...@sip.xxx.xx.xx. This correctly rated the call.

 So my next question is this, why does 0884567532 and rewritten correctly
 to 0061884567532 and 132221 not.

 What does the rewriting ?

 Thanks
 Mike

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So after thinking about my question for a little longer I realised I was
being very dim. There is an avp set which does this and I just needed to
make sure I was recording a value that CDRTool will rate ie make user it
starts with a 0.

Thanks for the help
Mike

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[OpenSIPS-Users] ERROR:perl:parser_init: failed to load perl file

2010-04-30 Thread samoh

Hi everybody,

I try to use the perl module to execute a script perl, I have compiled
opensips with this module and I put the path of my script in
modparam(perl, filename, /home/opensips/etc/opensips/script.pl) but it
doesn't work :(. I have these message in log file :

Apr 30 14:26:11 samy-desktop /home/opensips/sbin/opensips[21501]:
INFO:perl:mod_init: initializing... 
Apr 30 14:26:11 samy-desktop /home/opensips/sbin/opensips[21501]:
INFO:perl:parser_init: setting lib path: '/path/to/perllib' 
Apr 30 14:26:11 samy-desktop /home/opensips/sbin/opensips[21501]:
ERROR:perl:parser_init: failed to load perl file
/home/opensips/etc/opensips/script.pl. 
Apr 30 14:26:11 samy-desktop /home/opensips/sbin/opensips[21501]:
ERROR:core:init_mod: failed to initialize module perl 
Apr 30 14:26:11 samy-desktop /home/opensips/sbin/opensips[21501]:
ERROR:core:main: error while initializing modules 
Apr 30 14:29:01 samy-desktop opensips: INFO:core:init_tcp: using epoll_lt as
the TCP io watch method (auto detected) 
Apr 30 14:29:01 samy-desktop opensips: ERROR:core:daemonize: unable to
create pid file /var/run/opensips/opensips.pid: No such file or directory 

Someone can help me ?

Thanks.
Sam.

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Re: [OpenSIPS-Users] Modify Invite

2010-04-30 Thread Brad Bendy
Hi,

Im bringing this back up again.

Ive had to switch our INVITE format as some upstreams parse the INVITE
differently, I just need to change the order of variables around.

Now the weird thing is, I set the $ru and then when I check to see what
$ru is set to, it's been set correct, the order and all variables are
correct, but then openSIPs sends the INVITE out it fails to modify it at
all.

I was using:
$ru='sip:' + $rU + '@' + $rd + ';npdi=yes' + ';rn=' + $avp(s:lrn);

now
$ru='sip:' + $rU + ';npdi=yes' + ';rn=+1' + $avp(s:lrn) + '@' + $rd;

Before the INVITE would look like:

INVITE sip:+11235559...@1.1.1.1;npdi=yes;rn=+1123555 SIP/2.0

but now it should look like INVITE sip:+1123555;npdi=yes;rn=
+11235559...@1.1.1.1 SIP/2.0

it currently looks like INVITE sip:+11235559...@1.1.1.1 SIP/2.0.

Any ideals why just the positions of the AVPs and the other text would
matter? I think it's weird when I use xlog() to print what $ru is set to
it shows the correct value just OpenSIPs does not modify the INVITE. Not
sure if anyone else has ran into this on the list or not.

Thanks!


On Mon, 2010-02-22 at 13:21 -0700, Brad Bendy wrote:

 I just did: 
 $ru='sip:' + $rU + '@' + $rd + ';npdi=yes' + ';rn=' + $avp(s:lrn);
 
 That works like a champ, thanks for help on this, rewriting the ru
 direct took care of it!
 
 Thanks again
 
 On Mon, 2010-02-22 at 18:24 +0200, Bogdan-Andrei Iancu wrote: 
 
  Brad,
  
  the function does not support variables as parameters :(.
  
  But a simple workaround, if you want to add a parameter to the RURI is:
  $ru = $ru + ;rn= + $avp(s:foo) ;
  
  Regards,
  Bogdan
  
  Brad Bendy wrote:
   Hi Bogdan,
  
   I just tried the add_uri_param() but when a avp since I need to pass a 
   dynamic value to it, and it errors out with a syntax error. If I try 
   with double quotes around the avp it then literally displays 
   $avp(s:foo) when it adds to the URI.
  
   Something im missing?
  
   Thanks
  
   On Mon, 2010-02-22 at 10:45 +0200, Bogdan-Andrei Iancu wrote:
   Hi Brad,
  
   Maybe http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id270649 ?
  
   Regards,
   Bogdan
  
   Brad Bendy wrote:
Hi list,
   
I need to add npdi=yes;rn=xx when I send a INVITE out. Ive 
looked and don't really see any way to do this exactly. Is their a 
module im missing to do this, or some sort of method? I thought this 
would be in textops maybe but I do not see it in that module either.
   
Any help or pointers would be great.
   
Thanks!

   
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[OpenSIPS-Users] opensipsctl erros

2010-04-30 Thread Airton Kuada

Hi allThere is a sintaxe error in opensipsctl in lines 435, 438, 495, 498 when option is "opensipsctl alias_db show/list"if [ $DBENGINE -eq "MYSQL" ]if [ $DBENGINE -eq "PGSQL" ]toif [ $DBENGINE = "MYSQL" ]
if [ $DBENGINE = "PGSQL" ]
Airton Kuada--




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Re: [OpenSIPS-Users] opensipsctl erros

2010-04-30 Thread brianpocock

Just checked my configuration and it works fine for me with: 
if [ $DBENGINE -eq MYSQL ]
when doing opensipsctl alias_db list/show

Not sure whether you made a typo with:
if [ $DBENGINE -eq MYSQL ]
if [ $DBENGINE -eq PGSQL ]

but it should read:
if [ $DBENGINE -eq MYSQL ] 
elif [ $DBENGINE -eq PGSQL ] 

Hope this helps
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View this message in context: 
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Re: [OpenSIPS-Users] opensipsctl erros

2010-04-30 Thread Vadim Grinco
Hi,

On Fri, Apr 30, 2010 at 4:32 PM, brianpocock briankeithja...@googlemail.com
 wrote:


 Just checked my configuration and it works fine for me with:
 if [ $DBENGINE -eq MYSQL ]
 when doing opensipsctl alias_db list/show

 Not sure whether you made a typo with:
 if [ $DBENGINE -eq MYSQL ]
 if [ $DBENGINE -eq PGSQL ]

 but it should read:
 if [ $DBENGINE -eq MYSQL ]
 elif [ $DBENGINE -eq PGSQL ]

 I don't see any substantial difference between if and elif in this
situation. Yes, it will save you a couple of milliseconds to execute because
it won't do the second comparison, but the logyc is the same.
In this case the operator is incorrect. If you man sh, you will see that -eq
is an arithmetic binary operator, but = or == is used to compare strings.

You might have an interpreter which will understand both versions, but don't
forget there're lots of them out there.



 Hope this helps
 --
 View this message in context:
 http://opensips-open-sip-server.1449251.n2.nabble.com/opensipsctl-erros-tp4985896p4985961.html
 Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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-- 
Best regards,
Vadim Grinco
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[OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.conf doesn't exist

2010-04-30 Thread vishu gaddi
I have installed opensips 1.6, CDRTool 7.1.1 and RADIUS 1.1.7


To enable Opensips to send accounting logs to Freeradius, there should be
path defined in opensips.cfg as per documentation  as below:

*modparam(acc, aaa_url,radius:/etc/opensips/radius/client.conf)*



But there doesn't exist */etc/opensips/radius/client.conf* file, SO which
path i should define here.. i am suspected that my opensips is not compiled
with radius properly.. is it ?

Pls suggest..
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Re: [OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.conf doesn't exist

2010-04-30 Thread vishu gaddi
If directory  /etc/opensips/*radius* is generated at Opensips installation
time or Freeradius installation time ?

While installing Opensips, i had excluded aaa_radius from source file.


On Sat, May 1, 2010 at 3:24 AM, vishu gaddi vishuga...@gmail.com wrote:

 I have installed opensips 1.6, CDRTool 7.1.1 and RADIUS 1.1.7


 To enable Opensips to send accounting logs to Freeradius, there should be
 path defined in opensips.cfg as per documentation  as below:

 *modparam(acc, aaa_url,radius:/etc/opensips/radius/client.conf)*



 But there doesn't exist */etc/opensips/radius/client.conf* file, SO which
 path i should define here.. i am suspected that my opensips is not compiled
 with radius properly.. is it ?

 Pls suggest..

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Re: [OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.confdoesn't exist

2010-04-30 Thread duane . larson
Opensips is compiled and installed just fine.  You just haven't created the 
radius directory and moved a clients.conf file in there.
Sent via BlackBerry by ATT

-Original Message-
From: vishu gaddi vishuga...@gmail.com
Date: Sat, 1 May 2010 03:24:06 
To: Users@lists.opensips.org
Subject: [OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.conf
doesn't exist

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[OpenSIPS-Users] Trusted peer as a Prepaid client

2010-04-30 Thread Douglas Lane
Hi All,

I've been trying to get the following to work - am using OpenSIPS 
1.6.1-tls, and I've created a MySQL view of the address table tailored 
to what the old trusted table looked like.

Here is an example of the the invite:

Trusted Peer IP: 1.2.3.4
My OpenSIPS IP: 5.6.7.8

U 1.2.3.4:5060 - 5.6.7.8:5060
INVITE sip:0716054...@5.6.7.8:5060 SIP/2.0.
From: sip:0854003...@1.2.3.4;tag=1300083.
To: sip:0716054...@5.6.7.8:5060.
Call-ID: DCF50681-1BA8-47D2-82ED-DE5F0D91F596.
CSeq: 1 INVITE.
Privacy: none.
P-Asserted-Identity: tel:0854003684.
Max-Forwards: 69.
P-Charging-Vector: icid-value=91704882-49FC-4B8B-BACA-3E110481EE77.
User-Agent: vocl-essentra-ex/8.0F2 (19020.63).
Via: SIP/2.0/UDP 
1.2.3.4:5060;branch=z9hG4bK-8c76cf8-4bdb7875;vtservice=b2buaservlet.siptosip.
Contact: sip:0854003...@1.2.3.4:5060;vtservice=b2buaservlet.siptosip.
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK.
Content-Type: application/sdp.
Content-Length: 198.

Obviously this is handled correctly in the opensips routing logic, and 
the billing_party is set to 1.2.3.4 ($si), however, when call_control is 
called, the following message lands up in my syslog:

MaxSessionTime Duration=3600 CallId=22FBC00E-454E-4057-B835-15B527C774FD 
From=sip:0854003...@1.2.3.4 Gateway=1.2.3.4 To=sip:0716054...@5.6.7.8:5060
// Some of my debug after the MaxSessionTime call:
May  1 02:37:42 billing cdrtool[17219]: HACKING: 
[/var/www/CDRTool/library/cdr_opensips.php] 2452 aNumber = 
sip:0854003...@1.2.3.4
May  1 02:37:42 billing cdrtool[17219]: HACKING: 
[/var/www/CDRTool/library/cdr_opensips.php] 2465 BillingPartyId = 
0854003...@1.2.3.4
May  1 02:37:42 billing cdrtool[17219]: HACKING: select * from prepaid 
where account = '0854003...@1.2.3.4'
May  1 02:37:42 billing cdrtool[17219]: MaxSessionTime=unlimited 
Type=postpaid CallId=22FBC00E-454E-4057-B835-15B527C774FD 
billingparty=0854003...@1.2.3.4

Now I know call_control makes use of the From section to figure out the 
Billing Party ID, as demonstrated above. Also note the SQL query uses 
the full from portion, however, in my prepaid table, I only have 1.2.3.4 
as the customer (as the From user and display name section will change 
based on the CLID the trusted peer sends me)

I suppose the question is, what is the correct way (if any) to setup a 
trusted peer as a prepaid user where by the user section in the FROM 
header as well as the Display Name will consistently change. Only the 
domain part (in this case 1.2.3.4) will remain constant? I'd like to try 
avoid using uac_replace_from as I've heard this can mess things up for 
downstream / upstream proxies that my OpenSIPS might talk to eventually.

I would greatly appreciate the feedback on this.

Thanks
Doug


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