Opensips 1.6 as a Proxy and Presence Server

[maatohewetbi]

I use Opensips 1.6.2 on my server. I want to add BLF function to it but I don't know how. I have config where I route calls to asterisk. There are tutorials where I can set Opensip Presence Server as standalone, with OpenXcap as well, but there's nowhere config how to set Opensips to work on one machine as Presence Server, and also as a Proxy at same Opensips instance - in one config. How can I make it? Ok, I can set one Opensips as a proxy, and other Opensips (on another machine) as a Presence Server. But how can I connect them so that they work together?

[Răzvan Crainea-2]

Hello!

Basically merging the two separate configs together should do the trick. If you follow this[1] example, change your Proxy config as it follows:

When an initial INVITE comes in, create the dialog and publish the information:

	if (is_method("INVITE")) {
		create_dialog();
		# publish for both legs
		dialoginfo_set("AB");
	}

When a publish or subscribe methods come in handle them accordingly:

...
	if (is_method("PUBLISH|SUBSCRIBE")) {
		route(handle_presence);
		exit;
	}
...
route[handle_presence]
{
	if(!t_newtran()){
		sl_reply_error();
		exit;
	}

	if (is_method("PUBLISH")) {
		handle_publish();
	} else

	if (is_method("SUBSCRIBE")) {
		handle_subscribe();
	}

	exit;
}

[1] http://www.opensips.org/Documentation/Tutorials-Presence-PuaDialoinfoConfig PS: Please subscribe on the mailing list, otherwise you will loose further replies

On 01/03/2017 10:13 AM, maatohewetbi wrote:

I use Opensips 1.6.2 on my server. I want to add BLF function to it but I
don't know how. I have config where I route calls to asterisk. There are
tutorials where I can set Opensip Presence Server as standalone, with
OpenXcap as well, but there's nowhere config how to set Opensips to work on
one machine as Presence Server, and also as a Proxy at same Opensips
instance - in one config. How can I make it? Ok, I can set one Opensips as a
proxy, and other Opensips (on another machine) as a Presence Server. But how
can I connect them so that they work together? 



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[maatohewetbi]

It doesn't work. If I call number B, it rings, and BLF LED is blinking, but after picking up the phone, call is without audio and LED is still blinking event I disconnect call. Then If I try to call again, call freezes like if it's routing in loop, and B side doesn't ring at all. I heve to restart Opensips. Here is my config:

.................
Modules, general config
..................


# -------------------------  request routing logic -------------------
route{
    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    };
   
    if (msg:len >=  2048 ) {
        sl_send_reply("513", "Message too big");
        exit;
    };

    if (nat_uac_test("3")) {
        if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
            log("LOG:Someone trying to register from private IP, rewriting\n");
            fix_nated_contact();
            if ( is_method("INVITE") ) {
                fix_nated_sdp("1");
               };
            force_rport();
            setbflag(6);
            setbflag(8);
        };
    };

    if (loose_route()) {
        append_hf("P-hint: rr-enforced\r\n");
       
            if(src_ip == 'YYY.YYY.YYY.YYY'){
            route(1);
            }else{
            route(2);
            }
       
        exit;
    };

    if (!is_method("REGISTER"))
        record_route();

    if (!uri==myself) {
        append_hf("P-hint: outbound\r\n");
       
        if(src_ip == 'YYY.YYY.YYY.YYY'){
        route(1);
        }else{
        route(2);
        }
       
        exit;
    };

    if (uri==myself) {

        if (is_method("REGISTER")) {
            if (!www_authorize("", "subscriber")) {
                www_challenge("", "0");
                return;
            };
            save("location");
            exit;
        };

        lookup("aliases");
        if (!uri==myself) {
            append_hf("P-hint: outbound alias\r\n");
           
            if(src_ip == 'YYY.YYY.YYY.YYY'){
            route(1);
            }else{
            route(2);
            }
           
            exit;
        };
        alias_db_lookup("dbaliases","d");
        if (!lookup("location")) {
            route(2);
            exit;
        };
    };

       
        if (is_method("INVITE")) {
            create_dialog();
            dialoginfo_set();
    }
        if (is_method("PUBLISH|SUBSCRIBE")) {
        route(handle_presence);
        exit;
    }

       

    append_hf("P-hint: usrloc applied\r\n");
    if(src_ip == 'YYY.YYY.YYY.YYY'){
    route(1);
    }else{
    route(2);
    }
   
   
}

       
route[handle_presence]
{
    if(!t_newtran()){
        sl_reply_error();
        exit;
    }

    if (is_method("PUBLISH")) {
        handle_publish();
    } else

    if (is_method("SUBSCRIBE")) {
        handle_subscribe();
    }

    exit;
}

       

route[1]
{
    if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){
        sl_send_reply("479", "We don't forward to private IP addresses");
        exit;
    };

    if (isbflagset(6)) {
        force_rtp_proxy();
    };

    t_on_reply("1");

    if (!t_relay()) {
        sl_reply_error();
    };
}

route[2] {
    force_rtp_proxy();
    rewritehostport ("YYY.YYY.YYY.YYY:xx");
    t_relay();
}

onreply_route[1] {
    if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
        fix_nated_contact();
        force_rtp_proxy();
    } else if (nat_uac_test("1")) {
        fix_nated_contact();
    };
}

[Bogdan-Andrei Iancu-2]

Were your calls properly working before adding the BLF and presence
scripting ?

Regards,

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

On 03.01.2017 15:00, maatohewetbi wrote:

> It doesn't work. If I call number B, it rings, and BLF LED is blinking, but
> after picking up the phone, call is without audio and LED is still blinking
> event I disconnect call. Then If I try to call again, call freezes like if
> it's routing in loop, and B side doesn't ring at all. I heve to restart
> Opensips. Here is my config:
>
> .................
> Modules, general config
> ..................
>
>
> # -------------------------  request routing logic -------------------
> route{
>      if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
>      };
>      
>      if (msg:len >=  2048 ) {
> sl_send_reply("513", "Message too big");
> exit;
>      };
>
>      if (nat_uac_test("3")) {
> if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
>    log("LOG:Someone trying to register from private IP, rewriting\n");
>    fix_nated_contact();
>    if ( is_method("INVITE") ) {
> fix_nated_sdp("1");
>        };
>    force_rport();
>    setbflag(6);
>    setbflag(8);
> };
>      };
>
>      if (loose_route()) {
> append_hf("P-hint: rr-enforced\r\n");
>
>    if(src_ip == 'YYY.YYY.YYY.YYY'){
>    route(1);
>    }else{
>    route(2);
>    }
>
> exit;
>      };
>
>      if (!is_method("REGISTER"))
> record_route();
>
>      if (!uri==myself) {
> append_hf("P-hint: outbound\r\n");
>
> if(src_ip == 'YYY.YYY.YYY.YYY'){
> route(1);
> }else{
> route(2);
> }
>
> exit;
>      };
>
>      if (uri==myself) {
>
> if (is_method("REGISTER")) {
>    if (!www_authorize("", "subscriber")) {
> www_challenge("", "0");
> return;
>    };
>    save("location");
>    exit;
> };
>
> lookup("aliases");
> if (!uri==myself) {
>    append_hf("P-hint: outbound alias\r\n");
>
>    if(src_ip == 'YYY.YYY.YYY.YYY'){
>    route(1);
>    }else{
>    route(2);
>    }
>
>    exit;
> };
> alias_db_lookup("dbaliases","d");
> if (!lookup("location")) {
>    route(2);
>    exit;
> };
>      };
>
>
> if (is_method("INVITE")) {
>    create_dialog();
>    dialoginfo_set();
>      }
> if (is_method("PUBLISH|SUBSCRIBE")) {
> route(handle_presence);
> exit;
>      }
>
>
>
>      append_hf("P-hint: usrloc applied\r\n");
>      if(src_ip == 'YYY.YYY.YYY.YYY'){
>      route(1);
>      }else{
>      route(2);
>      }
>      
>      
> }
>
>
> route[handle_presence]
> {
>      if(!t_newtran()){
> sl_reply_error();
> exit;
>      }
>
>      if (is_method("PUBLISH")) {
> handle_publish();
>      } else
>
>      if (is_method("SUBSCRIBE")) {
> handle_subscribe();
>      }
>
>      exit;
> }
>
>
>
> route[1]
> {
>      if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
> !search("^Route:")){
> sl_send_reply("479", "We don't forward to private IP addresses");
> exit;
>      };
>
>      if (isbflagset(6)) {
> force_rtp_proxy();
>      };
>
>      t_on_reply("1");
>
>      if (!t_relay()) {
> sl_reply_error();
>      };
> }
>
> route[2] {
>      force_rtp_proxy();
>      rewritehostport ("YYY.YYY.YYY.YYY:xx");
>      t_relay();
> }
>
> onreply_route[1] {
>      if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
> fix_nated_contact();
> force_rtp_proxy();
>      } else if (nat_uac_test("1")) {
> fix_nated_contact();
>      };
> }
>
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-tp7605411p7605416.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

_______________________________________________
Users mailing list
[hidden email]
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[Marcin Zieliński]

That's right. All calls were working well before I add these lines. I noticed, that when I erase these lines 
if (is_method("PUBLISH|SUBSCRIBE")) {
    route(handle_presence);
    exit;
     }

problem is gone, and all calls goes smoothly.

Were your calls properly working before adding the BLF and presence scripting ?

Regards,

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

On 03.01.2017 15:00, maatohewetbi wrote:

It doesn't work. If I call number B, it rings, and BLF LED is blinking, but
after picking up the phone, call is without audio and LED is still blinking
event I disconnect call. Then If I try to call again, call freezes like if
it's routing in loop, and B side doesn't ring at all. I heve to restart
Opensips. Here is my config:

.................
Modules, general config
..................


# -------------------------  request routing logic -------------------
route{
     if (!mf_process_maxfwd_header("10")) {
    sl_send_reply("483","Too Many Hops");
    exit;
     };
          if (msg:len >=  2048 ) {
    sl_send_reply("513", "Message too big");
    exit;
     };

     if (nat_uac_test("3")) {
    if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
        log("LOG:Someone trying to register from private IP, rewriting\n");
        fix_nated_contact();
        if ( is_method("INVITE") ) {
        fix_nated_sdp("1");
                };
        force_rport();
        setbflag(6);
        setbflag(8);
    };
     };

     if (loose_route()) {
    append_hf("P-hint: rr-enforced\r\n");
    
        if(src_ip == 'YYY.YYY.YYY.YYY'){
        route(1);
        }else{
        route(2);
        }
    
    exit;
     };

     if (!is_method("REGISTER"))
    record_route();

     if (!uri==myself) {
    append_hf("P-hint: outbound\r\n");
    
    if(src_ip == 'YYY.YYY.YYY.YYY'){
    route(1);
    }else{
    route(2);
    }
    
    exit;
     };

     if (uri==myself) {

    if (is_method("REGISTER")) {
        if (!www_authorize("", "subscriber")) {
        www_challenge("", "0");
        return;
        };
        save("location");
        exit;
    };

    lookup("aliases");
    if (!uri==myself) {
        append_hf("P-hint: outbound alias\r\n");
    
        if(src_ip == 'YYY.YYY.YYY.YYY'){
        route(1);
        }else{
        route(2);
        }   
    
        exit;
    };
    alias_db_lookup("dbaliases","d");
    if (!lookup("location")) {
        route(2);
        exit;
    };
     };

    
    if (is_method("INVITE")) {
        create_dialog();
        dialoginfo_set();
     }
    if (is_method("PUBLISH|SUBSCRIBE")) {
    route(handle_presence);
    exit;
     }

    

     append_hf("P-hint: usrloc applied\r\n");
     if(src_ip == 'YYY.YYY.YYY.YYY'){
     route(1);
     }else{
     route(2);
     }
          }

    
route[handle_presence]
{
     if(!t_newtran()){
    sl_reply_error();
    exit;
     }

     if (is_method("PUBLISH")) {
    handle_publish();
     } else

     if (is_method("SUBSCRIBE")) {
    handle_subscribe();
     }

     exit;
}

    

route[1]
{
     if (uri=~[hidden email] &&
!search("^Route:")){
    sl_send_reply("479", "We don't forward to private IP addresses");
    exit;
     };

     if (isbflagset(6)) {
    force_rtp_proxy();
     };

     t_on_reply("1");

     if (!t_relay()) {
    sl_reply_error();
     };
}

route[2] {
     force_rtp_proxy();
     rewritehostport ("YYY.YYY.YYY.YYY:xx");
     t_relay();
}

onreply_route[1] {
     if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
    fix_nated_contact();
    force_rtp_proxy();
     } else if (nat_uac_test("1")) {
    fix_nated_contact();
     };
}




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[maatohewetbi]

Have You got any clue, why this is happnning?

[Răzvan Crainea-2]

It depends on where you added that block. Also, did you also add the
handle_presence route in your script?

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 01/09/2017 10:58 AM, maatohewetbi wrote:

> Have You got any clue, why this is happnning?
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-tp7605411p7605485.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
> Users mailing list
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[maatohewetbi]

Yes, I added this route. You can check my config, it's in this post.

[maatohewetbi]

Hello, can anybody help? I need blf to work.

[Răzvan Crainea-2]

I checked your script and you are not handling publish/subscribe for
sequential requests. You should try to move the route(handle_presence);
call for sequential requests too.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 01/17/2017 10:07 AM, maatohewetbi wrote:

> Hello, can anybody help? I need blf to work.
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-tp7605411p7605593.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
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[maatohewetbi]

Where are the sequential requests? Can You point it in my script?

[Bogdan-Andrei Iancu-2]

Hi,

It is a bit difficult to work with OpenSIPS without knowing at least the
SIP basics. I suggest to follow there webinars:
* Introduction to SIP : http://www.opensips.org/Documentation/Webinars#toc10
* Routing in SIP : http://www.opensips.org/Documentation/Webinars#toc12

A basic OpenSIPS script with Presence support is here:
http://www.opensips.org/Documentation/Tutorials-Presence-SimplePresConfig

Check the block in the "has_totag" - that is the part handling the
sequential requests.

Regards,

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

On 01/17/2017 10:38 AM, maatohewetbi wrote:

> Where are the sequential requests? Can You point it in my script?
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-tp7605411p7605598.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
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[maatohewetbi]

You are right Bogdan, I forgot about these requests. Anyway, thanks for help - I will look at webinars.

[maatohewetbi]

I want to ask what bolded values mean:

# -- core presence params --
modparam("presence","server_address","sip:sa@10.10.10.10:5060")


# -- pua and pua_dialoginfo parameters --
modparam("pua_dialoginfo", "presence_server", "sip:sa@10.10.10.10:5060")

I don't really know what to write there.

[maatohewetbi]

I think I found the reason BLF doesn't work. I made a test. I've erased presenity table. Then:
1. I called other user, led is blinking, and presentity table shows record with xml bodies with early state.
2. Then I pickup a call, the same record changes with confirmed state.
3. Then I end this call, the same record changes to terminated state.

So first call was ok. But after it I made second call, but it turned out that presentity table wasn't erased, and previous records exist. So when I make this second call, xml body contains previous xml, and this right one! I think presentity table should be erased after a call. Do You know the reason?

[Carlos Oliva]

Hi Marcin:

I had the same problem years ago with 1.6 version. In the first call BLF works but the second call fails. I solved it using dialog module to check the mumber of calls and not doing pua_set_publish if the mubers of calls are > 1

I think this problem was solved on 1.11 version, wich is suported until 7th of May 2017, 1.6 version has reached end of life on 2010

regards,

2017-01-20 8:58 GMT+01:00 maatohewetbi <[hidden email]>:

I think I found the reason BLF doesn't work. I made a test. I've erased
presenity table. Then:
1. I called other user, led is blinking, and presentity table shows record
with xml bodies with early state.
2. Then I pickup a call, the same record changes with confirmed state.
3. Then I end this call, the same record changes to terminated state.

So first call was ok. But after it I made second call, but it turned out
that presentity table wasn't erased, and previous records exist. So when I
make this second call, xml body contains previous xml, and this right one! I
think presentity table should be erased after a call. Do You know the
reason?




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[maatohewetbi]

How did You check it? You checked calls from this sip login?

[Carlos Oliva]

you can count calls from/to a sip user using dialog profiling

You can read about it in opensips book ( https://books.google.es/books?id=y2wdDAAAQBAJ&pg=PA202&lpg=PA202&dq=opensips+dialog+count+calls&source=bl&ots=LFFoLxC9Gf&sig=fe_67u5d7xSNXm2gXUqbSOmTznM&hl=es&sa=X&ved=0ahUKEwiD6s3dy9DRAhXEUhQKHRoCAtUQ6AEIOzAD#v=onepage&q=opensips%20dialog%20count%20calls&f=false )

To do it you create a profile with value and add the current dialog with the sip username as value. After it you get the profile size, if it is >1 you do not do pua_setpublish because there is another ongoing call.

But as I said this problem is solved in 1.11 version, 1.6 is very old.

2017-01-20 11:15 GMT+01:00 maatohewetbi <[hidden email]>:

How did You check it? You checked calls from this sip login?



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[Bogdan-Andrei Iancu-2]

Hi,

Those are SIP URIs to be used as contacts by the presence server (a SIP
URI pointing back - as address- to the presence server); in a Contact
URI, the username part is no relevant (as the contact is a reference to
the IP location of the device), so you can put whatever you want as
username.

Regards,

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

On 01/19/2017 01:03 PM, maatohewetbi wrote:

> I want to ask what bolded values mean:
>
> # -- core presence params --
> modparam("presence","server_address","sip:*sa*@10.10.10.10:5060")
>
>
> # -- pua and pua_dialoginfo parameters --
> modparam("pua_dialoginfo", "presence_server", "sip:*sa*@10.10.10.10:5060")
>
> I don't really know what to write there.
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-tp7605411p7605662.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
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> [hidden email]
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