[OpenSIPS-Users] Feature request - automatic string to lower domains
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Database Record
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
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] one way audio problem
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
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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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
<>___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Register Module GRUU Length Calculation Errors
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
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
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
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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 <>___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[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 <>___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using Dispatcher with destination as a host name
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] one way audio problem
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
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
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] one way audio problem
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
