[OpenSIPS-Users] Feature request - automatic string to lower domains

2013-05-07 Thread Daniel Goepp
Have you considered a function or flag that automatically sets the domain
on the RURI, From and To all lower case?

We have a work around of course, to just do this manually, but figured it
would be good to give some feed back.

Specifically we had issues regarding auth with multi-domain support.  In
the auth module we leave the first parameter null, which by default pulls
from the To header.  However, if a device sends caps in the domain, it
affect the auth (which is case-sensitive for realm).

Just a thought, with regards to compliance with DNS and domains in general,
which are not case sensitive.

-dg
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Re: [OpenSIPS-Users] Database Record

2013-05-07 Thread dpa
The version of Opensips?

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 4:52 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

Hello,

 

Please can you make it more clear or give me an example  for the function
acc_log_request("Some comment", "Some table");

 

onreply_route[2] {

   

   if (is_method("INVITE") && t_check_status("200") ) {

   

acc_log_request("coonect time ","acc"); ???

  }

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Tuesday, May 07, 2013 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

Hello

 

Try http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293991

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 2:01 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Database Record

 

Dears,

 

I am using acc module and I am inserting two rows for each call (INVITE,BYE)
,

I have a problem such I want to update the  record in acc table before the
calls ends which have a method = INVITE, and now that cant be done now
since opensips 

Does not insert the invite record before the calls ends 

 

How to let the invite  record inserted in database before calls ends

 

 

Please advice 

 

 

Khaled Chehab

Senior NGN Engineer

Description: icucall

Operations Office - Lebanon

Office: +961 1 515155 ext 300

Mobile  : +961 3 045212

E-mail: kche...@icucall.com

MSN ID :khalidche...@hotmail.com 

Skype: k_chehab 

Web Site: http://www.icucall.com

 http://www.allohi.com

 

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Re: [OpenSIPS-Users] Database Record

2013-05-07 Thread Aamir
Hi Khaled,

You need to study the cdr flag options, if I have time I will definitely post 
the appropriate parameter here.

--Aamir
--- Sent from My BlackBerry ---

-Original Message-
From: "dpa" 
Sender: users-boun...@lists.opensips.org
Date: Tue, 7 May 2013 14:06:15 
To: 'OpenSIPS users mailling list'
Reply-To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Database Record

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Re: [OpenSIPS-Users] one way audio problem

2013-05-07 Thread Aamir
It should not be under main route block, just place it outside main route block.

For e.g.

route {

...

}

Place it here...

Or else paste your config file here.

--Aamir
--- Sent from My BlackBerry ---

-Original Message-
From: sermj 2012 
Date: Tue, 7 May 2013 18:01:45 
To: aamir chougule; OpenSIPS users mailling 
list
Subject: Re: [OpenSIPS-Users] one way audio problem

Thanku very much for your prompt response,

iam new to opensips, please tell me where to add these lines in opensips.cfg

route[nat_check] {
if (client_nat_test("3")) {
force_rport();
fix_contact();
nat_keepalive();
}
}

The above lines i have added in opensips.cfg under Routing logic,
when i start opensips server,iam getting errors,

please help me.

Nandini


On Tue, May 7, 2013 at 2:30 PM, aamir chougule  wrote:

> Hi Nandini,
>
> The parameters and modules that you need to turn ON in your opensips.cfg
> file:
>
> loadmodule "nat_traversal.so"
>
> The above line load the module and the given below paragraph will set to
> test the parameters.
>
> route[nat_check] {
> if (client_nat_test("3")) {
> force_rport();
> fix_contact();
> nat_keepalive();
> }
> }
>
> Everytime you route a call first test the calls through the
> route(nat_check) which will fix all the NAT handling parameters.
>
> For e.g. if you are gonna route INVITE request then you need to do it like
> this:
>
> if(is_method("INVITE")) {
> route(invite_requests);
> exit;
> }
>
> route[invite_requests] {
> route(nat_check);
>
> if(!lookup("location")) {
> sl_send_reply("404", "User Not registered");
> exit;
> }
> t_on_reply("user_reply");
> t_relay();
> exit;
> }
>
> Its just an example that how I do it and always you can explore things and
> read the modules provided by OpenSIPS and upgrade yourself to use this
> server in all possible cases.
>
> Regards,
>
> Aamir Chougule
> Cell: 09167989111
>
>   --
>  *From:* Aamir 
>
> *To:* OpenSIPS users mailling list 
> *Sent:* Tuesday, 7 May 2013 1:58 PM
> *Subject:* Re: [OpenSIPS-Users] one way audio problem
>
> Hi Nandini,
>
> You actually need to turn on the nat_traversal module I guess, will pass
> you the parameters if I get time to do so.
>
> --Aamir
> --- Sent from My BlackBerry ---
>
> -Original Message-
> From: sermj 2012 
> Sender: users-boun...@lists.opensips.org
> Date: Tue, 7 May 2013 13:49:04
> To: 
> Reply-To: OpenSIPS users mailling list 
> Subject: [OpenSIPS-Users] one way audio problem
>
> ___
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>
>
>
>
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Re: [OpenSIPS-Users] Database Record

2013-05-07 Thread M.Khaled W Chehab
Hello,

 

Please can you make it more clear or give me an example  for the function
acc_log_request("Some comment", "Some table");

 

onreply_route[2] {

   

   if (is_method("INVITE") && t_check_status("200") ) {

   

acc_log_request("coonect time ","acc"); ???

  }

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of dpa
Sent: Tuesday, May 07, 2013 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] Database Record

 

Hello

 

Try http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293991

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 2:01 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Database Record

 

Dears,

 

I am using acc module and I am inserting two rows for each call (INVITE,BYE)
,

I have a problem such I want to update the  record in acc table before the
calls ends which have a method = INVITE, and now that cant be done now
since opensips 

Does not insert the invite record before the calls ends 

 

How to let the invite  record inserted in database before calls ends

 

 

Please advice 

 

 

Khaled Chehab

Senior NGN Engineer

Description: icucall

Operations Office - Lebanon

Office: +961 1 515155 ext 300

Mobile  : +961 3 045212

E-mail: kche...@icucall.com

MSN ID :khalidche...@hotmail.com 

Skype: k_chehab 

Web Site: http://www.icucall.com

 http://www.allohi.com

 

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Re: [OpenSIPS-Users] Register Module GRUU Length Calculation Errors

2013-05-07 Thread Vlad Paiu

Hello,

Thanks very much for the fixes, Tolga.
I've pushed them on OpenSIPS trunk, 1.9 and 1.8.

About the angle brackets, I see the RFC 5626 states that '

the URN will be
  encapsulated by angle brackets ("<" and">") when it is placed
  within the quoted string value of the "+sip.instance" Contact
  header field parameter.

'

But indeed, we could make the code more fault-tolerant for buggy clients 
by not assuming that the angle brackets are always there.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


On 04/30/2013 12:42 PM, Bogdan-Andrei Iancu wrote:

Hello Tolga,

Once again, thank you for the report and patch - Vlad, the maintainer 
of the GRUU code will take a look on this asap and make the fix.


Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 04/29/2013 02:34 AM, Tolga Tarhan wrote:

All,

I've discovered another bug in the register module's GRUU handling. 
This time, the issue is that an incorrect length is calculated for a 
temp GRUU before it is base64 encoded. This causes the GRUU to not 
match when it's decoded (since two extra characters of garbage get 
encoded on accident).


In the 1.9.0 source, the problem is in modules/registrar/reply.c on 
line 191. The temp GRUU is calculated as time_len + aor->len + 
instance->len + callid->len + 3, however, when instance is actually 
appended to the memory buffer, two characters (the leading and 
trailing angle brackets) are removed. This results in the reported 
length being two characters too long and two extra characters of 
garbage being included in the base64 encoded string.


I've created and verified a patch for this problem. It can be found 
here: 
http://netbrains-misc.s3.amazonaws.com/opensips/opensips-register-make-gruu-wrong-length.patch


Additionally, there appears to be a possibly related bug in 
modules/registrar/common.c on line 141 where the call id length after 
base64 decoding is inexplicably reduced by one. This may have been a 
previous attempt by someone to partially workaround the encoding bug 
above, but it isn't correct, as the last character of the GRUU call 
id is lost.


I've created and verified a patch for this as well. It can be found 
here: 
http://netbrains-misc.s3.amazonaws.com/opensips/opensips-lookup-gruu-wrong-length.patch


For what it it's worth, the assumption that sip.instance contains 
angle-brackets may be wrong. I believe that it's always supposed to, 
but assuming that it does is probably problematic and could be a 
source of even bigger problems if the instance is less than two 
characters long (where the memcpy would just grab random memory, I 
think). My patch doesn't address this aspect, however.


Please let me know if there's something else I need to do to get 
these patches accepted upstream.


Thanks,
Tolga Tarhan


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Re: [OpenSIPS-Users] one way audio problem

2013-05-07 Thread sermj 2012
Thanku very much for your prompt response,

iam new to opensips, please tell me where to add these lines in opensips.cfg

route[nat_check] {
if (client_nat_test("3")) {
force_rport();
fix_contact();
nat_keepalive();
}
}

The above lines i have added in opensips.cfg under Routing logic,
when i start opensips server,iam getting errors,

please help me.

Nandini


On Tue, May 7, 2013 at 2:30 PM, aamir chougule  wrote:

> Hi Nandini,
>
> The parameters and modules that you need to turn ON in your opensips.cfg
> file:
>
> loadmodule "nat_traversal.so"
>
> The above line load the module and the given below paragraph will set to
> test the parameters.
>
> route[nat_check] {
> if (client_nat_test("3")) {
> force_rport();
> fix_contact();
> nat_keepalive();
> }
> }
>
> Everytime you route a call first test the calls through the
> route(nat_check) which will fix all the NAT handling parameters.
>
> For e.g. if you are gonna route INVITE request then you need to do it like
> this:
>
> if(is_method("INVITE")) {
> route(invite_requests);
> exit;
> }
>
> route[invite_requests] {
> route(nat_check);
>
> if(!lookup("location")) {
> sl_send_reply("404", "User Not registered");
> exit;
> }
> t_on_reply("user_reply");
> t_relay();
> exit;
> }
>
> Its just an example that how I do it and always you can explore things and
> read the modules provided by OpenSIPS and upgrade yourself to use this
> server in all possible cases.
>
> Regards,
>
> Aamir Chougule
> Cell: 09167989111
>
>   --
>  *From:* Aamir 
>
> *To:* OpenSIPS users mailling list 
> *Sent:* Tuesday, 7 May 2013 1:58 PM
> *Subject:* Re: [OpenSIPS-Users] one way audio problem
>
> Hi Nandini,
>
> You actually need to turn on the nat_traversal module I guess, will pass
> you the parameters if I get time to do so.
>
> --Aamir
> --- Sent from My BlackBerry ---
>
> -Original Message-
> From: sermj 2012 
> Sender: users-boun...@lists.opensips.org
> Date: Tue, 7 May 2013 13:49:04
> To: 
> Reply-To: OpenSIPS users mailling list 
> Subject: [OpenSIPS-Users] one way audio problem
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
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>
>
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>
>
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Re: [OpenSIPS-Users] Database Record

2013-05-07 Thread dpa
Hello

 

Try http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293991

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Tuesday, May 07, 2013 2:01 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Database Record

 

Dears,

 

I am using acc module and I am inserting two rows for each call (INVITE,BYE)
,

I have a problem such I want to update the  record in acc table before the
calls ends which have a method = INVITE, and now that cant be done now
since opensips 

Does not insert the invite record before the calls ends 

 

How to let the invite  record inserted in database before calls ends

 

 

Please advice 

 

 

Khaled Chehab

Senior NGN Engineer

Description: icucall

Operations Office - Lebanon

Office: +961 1 515155 ext 300

Mobile  : +961 3 045212

E-mail: kche...@icucall.com

MSN ID :khalidche...@hotmail.com 

Skype: k_chehab 

Web Site: http://www.icucall.com

 http://www.allohi.com

 

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[OpenSIPS-Users] Database Record

2013-05-07 Thread M.Khaled W Chehab
Dears,

 

I am using acc module and I am inserting two rows for each call (INVITE,BYE)
,

I have a problem such I want to update the  record in acc table before the
calls ends which have a method = INVITE, and now that cant be done now
since opensips 

Does not insert the invite record before the calls ends 

 

How to let the invite  record inserted in database before calls ends

 

Please advice 

 

 

Khaled Chehab

Senior NGN Engineer

Description: icucall

Operations Office - Lebanon

Office: +961 1 515155 ext 300

Mobile  : +961 3 045212

E-mail: kche...@icucall.com

MSN ID :khalidche...@hotmail.com 

Skype: k_chehab 

Web Site: http://www.icucall.com

 http://www.allohi.com

 

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[OpenSIPS-Users] Using Dispatcher with destination as a host name

2013-05-07 Thread John Quick
Sorry, I just realised there may be other factors in this duplicate INVITE
problem so it may not be connected with the DNS resolving to two IP's.
I would still be interested to know how dispatcher should behave when using
a host name that resolves to more than one IP.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk




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Re: [OpenSIPS-Users] one way audio problem

2013-05-07 Thread aamir chougule
Hi Nandini,

The parameters and modules that you need to turn ON in your opensips.cfg file:

loadmodule "nat_traversal.so"

The above line load the module and the given below paragraph will set to test 
the parameters.


route[nat_check] {
    if (client_nat_test("3")) {
    force_rport();
    fix_contact();
    nat_keepalive();
    }
}

Everytime you route a call first test the calls through the route(nat_check) 
which will fix all the NAT handling parameters.

For e.g. if you are gonna route INVITE request then you need to do it like this:

    if(is_method("INVITE")) {
    route(invite_requests);
    exit;
    }


route[invite_requests] {
    route(nat_check);

    if(!lookup("location")) {
    sl_send_reply("404", "User Not registered");
    exit;
    }
    t_on_reply("user_reply");
    t_relay();
    exit;
    }


Its just an example that how I do it and always you can explore things and read 
the modules provided by OpenSIPS and upgrade yourself to use this server in all 
possible cases.
 
Regards,

Aamir Chougule
Cell: 09167989111




 From: Aamir 
To: OpenSIPS users mailling list  
Sent: Tuesday, 7 May 2013 1:58 PM
Subject: Re: [OpenSIPS-Users] one way audio problem
 

Hi Nandini,

You actually need to turn on the nat_traversal module I guess, will pass you 
the parameters if I get time to do so.

--Aamir
--- Sent from My BlackBerry ---

-Original Message-
From: sermj 2012 
Sender: users-boun...@lists.opensips.org
Date: Tue, 7 May 2013 13:49:04 
To: 
Reply-To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] one way audio problem

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[OpenSIPS-Users] Using Dispatcher with destination as a host name

2013-05-07 Thread John Quick
I am using the Dispatcher module on OpenSIPS version 1.8.2-notls.
The destination field for 'Set 2' is sip:sipipgw.magrathea.net (which
resolves in DNS to two different IP addresses).
The code in my failure_route when a call has failed to reach a 'Set 1'
destination, is essentially like this:
ds_select_domain("2", "0");
route(15);

...and route 15 contains some lines like this:
t_on_failure("3");
t_relay();

After executing the code outlined above, a SIP trace shows that OpenSIPS
sends two identical INVITE requests to one magrathea address. The two
INVITE's are sent almost simultaneously. Is this an error? I would expect to
see one INVITE to one address or maybe one to each address (parallel
forking), but not two to one address.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk




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Re: [OpenSIPS-Users] one way audio problem

2013-05-07 Thread Aamir
Hi Nandini,

You actually need to turn on the nat_traversal module I guess, will pass you 
the parameters if I get time to do so.

--Aamir
--- Sent from My BlackBerry ---

-Original Message-
From: sermj 2012 
Sender: users-boun...@lists.opensips.org
Date: Tue, 7 May 2013 13:49:04 
To: 
Reply-To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] one way audio problem

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[OpenSIPS-Users] one way audio problem

2013-05-07 Thread sermj 2012
i configured and installed opensips successfully,i have registerd two
clients:-

6005 :- 192.168.2.48
6006:- 192.168.2.50

i can hear only one way audio.iam using wifi standalone router to
communicate with clients.i have searched in the blogs to slove the problem.
by seeing the blogs i came to know that rtptproxy would slove problem.

i have integrated opensips with rtpproxy module successfully,
but still the problem is not sloved.

iam attaching the sip trace file and opensips.cfg file.

please help me from this issue.

Thank you
Nandini


opensips.cfg
Description: Binary data


sip
Description: Binary data
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