date:20210311

[OpenSIPS-Users] OpenSIPs & CP - Documentation (Dynamic Routing, DialPlans, CP)

2021-03-11 Thread Rick McGill - ₪

Dear OpenSIPs Community,

I'm very new to OpenSIPS and have a new fresh install and have started to
customize it.
So if my question seems so very basic this is why.

I have the following:
Debian 1.7 (Buster)
OpenSIPs 3.1
OpenSIPs Control Panel
RTPproxy

I have a couple of uses setup and can call with softphone from User to user
on OpenSIPS.
I have an FreePBX/Asterisk SIP Trunk setup to OpenSIPS.   With outbound
routes and all configured and working properly.  
I also have on OpenSIPS a Route/Gateway or Trunk from OpenSIPs to CommPeak
For Outbound calls.

The issue I'm having is learning or finding documentation on how exactly to
setup Dialplan, Dynamic Routes, Permissions and such so that I can get a
call from Ext on Asterisk routed to OpenSIPS then out to CommPeak for an
outbound International call.

Currently when I try to make this type of call OpenSIPS says for example
662...@sip.netrovoip.comuserloc does not exists and hangs up.
Sip.netrovoip.com is my OpenSIPs server
So apparently with all my efforts my Dialplan, Permissions, and Dynamic
Routes is not working no matter what changes I keep trying in them.

FYI I have not yet edited or customized the file
/var/www/opensips-cp/config/tools/system/dialplan/local.inc.php

I cannot find out what code and customization I should do for attrib and
what I can use standard in OpenSIPS-CP

$config->attrs_cb=array(
<-->// name , description
<-->"a" => "Descr a",
<-->"b" => "Descr b",
<-->"c" => "Descr c",

So My question is:
Where can I find some comprehensive detailed documentation (if it exists) so
I can learn more about the following?:
Dialplans
Dynamic Routes
And whatever else I need to learn to get this kind of basic Dynamic Routing
working?
I have gone thru the OpenSIPS-CP documentation and the OpenSIPs.org too but
there it seems a bit basic or outdated.

Also I do have my Regular Expressions Quick Reference v1.2 but no matter
what I program in CP it does not seem to work.
I think I'm missing some very basic but major bit of information on how to
get this simple Asterisk -> OpenSIPS -> CommPeak outbound call basic setup
configured / working.

Again sorry if all this sound like such a basic knowledge set of questions.


---
Rick McGill – CEO
r...@netrovoip.com | r...@netropolitanworks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |Mobile:
+66-85557-3000
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  |
supp...@netrovoip.com
  ₪  www.NetroVOIP.com  Telecommunications / Video Consulting & Solutions
Provider

---   



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Re: [OpenSIPS-Users] Opensips 3.1 as HEP server

2021-03-11 Thread Saurabh Chopra

Hi All,

Kindly look into this problem and please update us with your respected
thoughts

Best Regards
Saurabh Chopra
+918861979979


On Wed, Mar 10, 2021 at 6:15 PM Saurabh Chopra  wrote:

> Hi All,
>
> I am trying to configure a HEP server on Opensips 3.1.0 but while testing
> my server is crashing abruptly.
>
> I am following the configuration defined under section 6.1 Homer5
> integration on " https://www.opensips.org/Documentation/Tutorials-Tracing
> "
>
> Below is the sample configuration at my opensips server.
>
> log_level=4
> log_stderror=no
> log_facility=LOG_LOCAL5
> socket=hep_udp:104.131.xx.xx:6xxx
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> *loadmodule "proto_hep.so"loadmodule "sipcapture.so"modparam("sipcapture",
> "db_url", "mysql://:x@localhost/homerdb")modparam("sipcapture",
> "hep_capture_on", 1)modparam("sipcapture", "hep_route", "HEPR")###
> Routing Logic # main request routing logicroute{if($rm =~
> "REGISTER") {$var(table) = "sip_capture_registration";
>   }else if($rm =~ "(INVITE|UPDATE|BYE|ACK|PRACK|REFER|CANCEL)$")
> {$var(table) = "sip_capture_call";}else
> if($rm =~ "(NOTIFY)$" && is_present_hf("Event") && $hdr(Event)=~"refer;")
>   {$var(table) = "sip_capture_call";}
> else if($rm =~ "(INFO)$"){$var(table) =
> "sip_capture_call";}else if($rm =~ "(OPTIONS)$" ){
>   $var(table) = "sip_capture_rest";}else {
>   $var(table) = "sip_capture_rest";}#$var(utc) = "%Y%m%d";
>   xlog("WRITING NEWLY CAME PACKET INTO ($var(table))!\n");
> if($var(table) == "sip_capture_call")
> sip_capture("sip_capture_call_%Y%m%d");else if($var(table) ==
> "sip_capture_registration")
> sip_capture("sip_capture_registration_%Y%m%d");else
> sip_capture("sip_capture_rest_%Y%m%d");}route[HEPR] {
> hep_get("11", $var(vid), $var(data));if ( $var(data) == "SIP" ) {
>   hep_resume_sip();}$var(proto) = $(var(data){s.int
> });### anything but xlog shall be droppedif (
> $var(proto) != 87 )exit;hep_get("0x0011",
> "utf8-string", $var(vendor), $var(correlation));
> report_capture($var(correlation_id), "rtcp_log", $var(proto_type));
> #report_capture("logs_capture", $var(correlation));exit;}*
>
> Any prompt help would be highly appreciated.
>
> Best Regards
> Saurabh Chopra
> +918861979979
>
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[OpenSIPS-Users] Changing the expire parameter in the Contact field

2021-03-11 Thread Joseph Barrero

Is there a way to replace the expires parameter in a Contact header within
a reply route?

The issue I'm trying to solve is the following.

I have two OpenSIPS servers, a primary and a backup.  The OpenSIPS server
is performing mid-registrar duties for Freeswitch.  However, when I take
the primary OpenSIPS server down for maintenance and bring it back online,
Freeswitch still has the other registration active.  When the UA tries to
register again on the primary OpenSIPS server, the mid-register module
sends a request over to Freeswitch, and Freeswitch returns two contacts;
one contact with the modparam("mid_registrar", "outgoing_expires", 600)
value and the other contact with an expiration of whatever value it happens
to be on (in this example it is 600 - time elapsed) from the registration
created by the secondary OpenSIPS server.

The problem is that the expiration parameter in the 200 Ok reply to the UA
from OpenSIPS on a successful registration has the second expiration value
instead of the shorter expiration value set in modparam("mid_registrar",
"max_expires", 60).

The result, is that the UA goes from 60-second registrations to whatever
the second expiration value had at the time (a value between 0 and 599).
When the second expiration is higher than 60 seconds, the UA's registration
is expired on OpenSIPS for up to 599 seconds.

What I'd like to do is simply change the expire parameter in the contact
header to 60 seconds on the reply route.

Any help will be greatly appreciated.

Thank you,
Joe Barrero
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Re: [OpenSIPS-Users] How to manage REFER Packet

2021-03-11 Thread johan

302 you mean ?

On 11/03/2021 17:24, Calvin Ellison wrote:
> Is it your intention to transfer an established call, or do you really
> a redirect?
>
> In the case of a redirect you're actually looking for the REDIREDCT
> method not the REFER method. Instead of returning a 200 OK then REFER,
> you would return a 300 REDIRECT with the new URI in the Contact
> header. That completes the first dialogue and then you start a new
> dialogue to the destination provided by the REDIRECT.
>
>
> Regards,
>
>
> Calvin Ellison
> Systems Architect
> calvin.elli...@voxox.com 
> +1 (213) 285-0555
>
> ---
> voxox.com  
> 5825 Oberlin Drive, Suite 5
> San Diego, CA 92121
>
> On Sun, Mar 7, 2021, 23:12 Vinayak Makwana
> mailto:vinayak.makw...@ecosmob.com>> wrote:
>
> Hello Everyone,
>  I would like to manage REFER packet with opensips script 
>  In my case i want manage REFER packet like whenever i opensips
> proxy getting REFER packet from freeswitch at that time i need to
> convert this REFER packet into INVITE packet from opensips script
> and send to the endpoint.
> Is their  any possibilities where I can manage this thing in
> opensips 3.1.x ?
> Please suggest me.
>
> My scenario like this:
>
> A                            opensips                B(Freswitch)
>   |   -INVITE --->    |   ---INVITE-->   |       
>                      
>   |   <-200OK ---    |  <---200OK--    | 
>   |  <-INVITE ---   |   <---REFER    |
>
> *Disclaimer*
> In addition to generic Disclaimer which you have agreed on our
> website, any views or opinions presented in this email are solely
> those of the originator and do not necessarily represent those of
> the Company or its sister concerns. Any liability (in negligence,
> contract or otherwise) arising from any third party taking any
> action, or refraining from taking any action on the basis of any
> of the information contained in this email is hereby excluded.
>
> *Confidentiality*
> This communication (including any attachment/s) is intended only
> for the use of the addressee(s) and contains information that is
> PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination,
> distribution, or copying of this communication is prohibited.
> Please inform originator if you have received it in error.
>
> *Caution for viruses, malware etc.*
> This communication, including any attachments, may not be free of
> viruses, trojans, similar or new contaminants/malware,
> interceptions or interference, and may not be compatible with your
> systems. You shall carry out virus/malware scanning on your own
> before opening any attachment to this e-mail. The sender of this
> e-mail and Company including its sister concerns shall not be
> liable for any damage that may incur to you as a result of
> viruses, incompleteness of this message, a delay in receipt of
> this message or any other computer problems. 
> ___
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>
>
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Re: [OpenSIPS-Users] How to manage REFER Packet

2021-03-11 Thread Donat Zenichev

Good day Calvin.
There is no SIP method called "REDIREDCT" as you say.
Currently there are 6 general methods defined in RFC 3261, you can see them
here: https://tools.ietf.org/html/rfc3261#section-27.4

And of course a list of SIP specifications (documents) including following
methods: PRACK, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, UPDATE
which are defined in the list of other RFC documents.

But there is no method which you called. So that, please do not confuse the
community with a wrong terminology.

The thing you are referring to (if I managed to understand this correctly)
is just a basic processing of 3XX type of responses (Redirection server).
This block of response codes is commonly called as redirection responses.
Response codes are not the same thing as SIP methods.
And most of all would be "302 Moved Temporarily" used, to provide Contact
information where to find the requested side. Depends on setup.

This however seems to be completely not related to the original question of
the author.

Best regards.


On Thu, Mar 11, 2021 at 6:26 PM Calvin Ellison 
wrote:

> Is it your intention to transfer an established call, or do you really a
> redirect?
>
> In the case of a redirect you're actually looking for the REDIREDCT method
> not the REFER method. Instead of returning a 200 OK then REFER, you would
> return a 300 REDIRECT with the new URI in the Contact header. That
> completes the first dialogue and then you start a new dialogue to the
> destination provided by the REDIRECT.
>
>
> Regards,
>
>
> Calvin Ellison
> Systems Architect
> calvin.elli...@voxox.com
> +1 (213) 285-0555
>
> ---
> voxox.com
> 5825 Oberlin Drive, Suite 5
> San Diego, CA 92121
>
> On Sun, Mar 7, 2021, 23:12 Vinayak Makwana 
> wrote:
>
>> Hello Everyone,
>>  I would like to manage REFER packet with opensips script
>>  In my case i want manage REFER packet like whenever i opensips proxy
>> getting REFER packet from freeswitch at that time i need to convert this
>> REFER packet into INVITE packet from opensips script and send to the
>> endpoint.
>> Is their  any possibilities where I can manage this thing in opensips
>> 3.1.x ?
>> Please suggest me.
>>
>> My scenario like this:
>>
>> AopensipsB(Freswitch)
>>   |   -INVITE --->|   ---INVITE-->   |
>>
>>   |   <-200OK ---|  <---200OK--|
>>   |  <-INVITE ---   |   <---REFER|
>>
>> *Disclaimer*
>> In addition to generic Disclaimer which you have agreed on our website,
>> any views or opinions presented in this email are solely those of the
>> originator and do not necessarily represent those of the Company or its
>> sister concerns. Any liability (in negligence, contract or otherwise)
>> arising from any third party taking any action, or refraining from taking
>> any action on the basis of any of the information contained in this email
>> is hereby excluded.
>>
>> *Confidentiality*
>> This communication (including any attachment/s) is intended only for the
>> use of the addressee(s) and contains information that is PRIVILEGED AND
>> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying
>> of this communication is prohibited. Please inform originator if you have
>> received it in error.
>>
>> *Caution for viruses, malware etc.*
>> This communication, including any attachments, may not be free of
>> viruses, trojans, similar or new contaminants/malware, interceptions or
>> interference, and may not be compatible with your systems. You shall carry
>> out virus/malware scanning on your own before opening any attachment to
>> this e-mail. The sender of this e-mail and Company including its sister
>> concerns shall not be liable for any damage that may incur to you as a
>> result of viruses, incompleteness of this message, a delay in receipt of
>> this message or any other computer problems.
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
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>


-- 

Best regards,
Donat Zenichev
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[OpenSIPS-Users] Compact SIP

2021-03-11 Thread Saint Michael

>
> I needd to send out only compact sip. I use Opensips. 2.4. Can somebody
> indicate how do I change my configuration to achieve 100% compact sip? I
> found some information but I am afraid is not enough for a succesful
> implementation..

Philip
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Re: [OpenSIPS-Users] How to manage REFER Packet

2021-03-11 Thread Calvin Ellison

Is it your intention to transfer an established call, or do you really a
redirect?

In the case of a redirect you're actually looking for the REDIREDCT method
not the REFER method. Instead of returning a 200 OK then REFER, you would
return a 300 REDIRECT with the new URI in the Contact header. That
completes the first dialogue and then you start a new dialogue to the
destination provided by the REDIRECT.


Regards,


Calvin Ellison
Systems Architect
calvin.elli...@voxox.com
+1 (213) 285-0555

---
voxox.com
5825 Oberlin Drive, Suite 5
San Diego, CA 92121

On Sun, Mar 7, 2021, 23:12 Vinayak Makwana 
wrote:

> Hello Everyone,
>  I would like to manage REFER packet with opensips script
>  In my case i want manage REFER packet like whenever i opensips proxy
> getting REFER packet from freeswitch at that time i need to convert this
> REFER packet into INVITE packet from opensips script and send to the
> endpoint.
> Is their  any possibilities where I can manage this thing in opensips
> 3.1.x ?
> Please suggest me.
>
> My scenario like this:
>
> AopensipsB(Freswitch)
>   |   -INVITE --->|   ---INVITE-->   |
>
>   |   <-200OK ---|  <---200OK--|
>   |  <-INVITE ---   |   <---REFER|
>
> *Disclaimer*
> In addition to generic Disclaimer which you have agreed on our website,
> any views or opinions presented in this email are solely those of the
> originator and do not necessarily represent those of the Company or its
> sister concerns. Any liability (in negligence, contract or otherwise)
> arising from any third party taking any action, or refraining from taking
> any action on the basis of any of the information contained in this email
> is hereby excluded.
>
> *Confidentiality*
> This communication (including any attachment/s) is intended only for the
> use of the addressee(s) and contains information that is PRIVILEGED AND
> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying
> of this communication is prohibited. Please inform originator if you have
> received it in error.
>
> *Caution for viruses, malware etc.*
> This communication, including any attachments, may not be free of viruses,
> trojans, similar or new contaminants/malware, interceptions or
> interference, and may not be compatible with your systems. You shall carry
> out virus/malware scanning on your own before opening any attachment to
> this e-mail. The sender of this e-mail and Company including its sister
> concerns shall not be liable for any damage that may incur to you as a
> result of viruses, incompleteness of this message, a delay in receipt of
> this message or any other computer problems.
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] OpenSIPs & CP - Documentation (Dynamic Routing, DialPlans, CP)

Re: [OpenSIPS-Users] Opensips 3.1 as HEP server

[OpenSIPS-Users] Changing the expire parameter in the Contact field

Re: [OpenSIPS-Users] How to manage REFER Packet

Re: [OpenSIPS-Users] How to manage REFER Packet

[OpenSIPS-Users] Compact SIP

Re: [OpenSIPS-Users] How to manage REFER Packet

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