[OpenSIPS-Users] Users Digest, Vol 129, Issue 20
Angel Fernández Sánchez
angelfernandezsanchez at gmail.com
Wed Apr 24 11:22:22 EDT 2019
El lun., 22 abr. 2019 a las 1:24, <users-request at lists.opensips.org>
escribió:
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> Today's Topics:
>
> 1. Re: Fwd: Opensips SNGTC Module (Liviu Chircu)
> 2. Re: WebRTC and mid_registrar issue (Liviu Chircu)
> 3. Re: Integrating with Asterisk in the same box (Liviu Chircu)
> 4. Re: async() and config actions call stack (Liviu Chircu)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 22 Apr 2019 09:07:00 +0300
> From: Liviu Chircu <liviu at opensips.org>
> To: users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] Fwd: Opensips SNGTC Module
> Message-ID: <137f8a9a-774e-b381-3b20-97dd6a23bb77 at opensips.org>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
> Hi Angel,
>
> Sorry for missing out on this one -- the sngtc_server daemon should be
> aware of all cards present on the network (IIRC, it auto-detects them).
> Similarly, the sngtc library and OpenSIPS module will auto-detect
> sngtc_server and make requests to it.
>
> It may sound like too good to be true, but it's how it actually works --
> that's how I developed/tested the module in the first place.
>
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
Ok, thanks very much.
Regarding the method sngtc_callee_answer([listen_if_A[, listen_if_B]])
¿Are listen_if_A and listen_if_B optional parameters?
If not, what are the values of the IPs to use?
Best Regards
>
>
> On 05.04.2019 20:15, Angel Fernández Sánchez wrote:
> >
> >
> > Hello all,
> >
> > I'm trying to test the SNGTC module.
> > I have setup an installation of opensips in a machine and a sngtc
> > server in the same machine, in port 9000 with a sangoma D150
> > configured in it which is connected to the same LAN via ethernet port.
> >
> > I get no clear idea about how to specify to sngtc module how to
> > connect to the sngtc server. ¿Should I hardcode server data in the
> > library and compile?
> >
> > Could you help me with this test configuration?
> >
> > Best regards.
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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> ------------------------------
>
> Message: 2
> Date: Mon, 22 Apr 2019 09:10:54 +0300
> From: Liviu Chircu <liviu at opensips.org>
> To: users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] WebRTC and mid_registrar issue
> Message-ID: <25c7fc75-c2e9-a768-3976-7db5edd35937 at opensips.org>
> Content-Type: text/plain; charset="windows-1252"; Format="flowed"
>
> Hi Terry,
>
>
> Currently, mid-registrar is incompatible with topology hiding, as they
> both attempt to edit the "Contact" header field.
>
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
> On 09.04.2019 14:01, Terry Walters wrote:
> > I have a working OpenSIPS deployment running as a PSTN proxy and am
> > trying to add in support for WebRTC which will forward requests to an
> > internal SIP PBX. Currently the registrations appear to be passing
> > through the mid_registrar correctly and the initial invite from my PBX
> > is correctly routed to the websocket client.
> >
> > The PBX receives messages up to the 180 ringing correctly, but on the
> > 200 OK from the websocket client the r.uri contains a d.id rather than
> > the ctid, this then results in the calling timing out and the ACK from
> > the PBX not routing to the websocket client. I believe this is the
> > script area that is falling over once the has_totag function is checked.
> >
> > if (loose_route())
> > {
> > if (is_method("INVITE"))
> > {
> > # even if in most of the cases is useless, do RR for
> > # re-INVITEs alos, as some buggy clients do change route set
> > # during the dialog.
> > record_route();
> > }
> > else if (is_method("ACK"))
> > {
> > if (has_body("application/sdp"))
> > {
> > # check if destination is WS
> > if ($du != NULL)
> > $var(proto) = $dP;
> > else
> > $var(proto) = $rP;
> > if ($var(proto) == "WS" || $var(proto) == "WSS")
> > setbflag(DST_WS);
> >
> > route(rtpengine_answer);
> > }
> > }
> >
> > # route it out to whatever destination was set by loose_route()
> > # in $du (destination URI).
> > route(websocket);
> > }
> > else
> > {
> > if ( is_method("ACK") )
> > {
> > if ( t_check_trans() )
> > {
> > # non loose-route, but stateful ACK; must be an ACK after
> > # a 487 or e.g. 404 from upstream server
> > t_relay();
> > exit;
> > }
> > else
> > {
> > # ACK without matching transaction ->
> > # ignore and discard
> > exit;
> > }
> > }
> > sl_send_reply("404","Not here");
> >
> > The route[websocket] is using force send socket to remove the NAT
> > address that cannot be accessed internally:
> >
> > route[webscoket]
> > {
> > xlog("L_INFO","Entered websocket route");
> > # for each branch we will call the function below
> > t_on_branch("per_branch_ops");
> >
> > # for each reply we will call the function below
> > t_on_reply("handle_nat");
> > # initial invites from the main registrar - need to look them up!
> > if (is_method("INVITE"))
> > if($Ri == "1.1.1.1" and $Rp == 5070)
> > {
> > xlog("L_INFO","Call received for a websocket client");
> > if (!mid_registrar_lookup("location"))
> > {
> > t_reply("404", "Not Found");
> > exit;
> > }
> >
> > }
> > else if($Ri=="1.1.1.1" && $Rp == "8080")
> > {
> > xlog("L_INFO","Inbound call received on port 8080");
> > route("ToInternal");
> > t_on_failure("int_invites");
> > }
> >
> > if((is_method("REGISTER"))&&($Ri=="1.1.1.1")&&($Rp =="8080"))
> > {
> > xlog("L_INFO","Register received - mid-registrar actions");
> > fix_nated_register();
> > mid_registrar_save("location");
> > switch ($retcode)
> > {
> > case 1:
> > xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
> > $ru = "sip:2.2.2.2:5060";
> > force_send_socket(UDP:1.1.1.1:5070);
> > if (!t_relay()) {
> > send_reply("500", "Server Internal Error 1");
> > }
> > t_on_failure("int_invites");
> > break;
> > case 2:
> > xlog("L_INFO", "REGISTER has been absorbed!\n");
> > break;
> > default:
> > xlog("L_ERR", "mid-registrar error!\n");
> > send_reply("500", "Server Internal Error 2");
> > }
> >
> > exit;
> > }
> >
> > # removing the rtpproxy session
> > if(is_method("CANCEL|BYE"))
> > {
> > rtpengine_delete();
> > }
> >
> > # try to send the request on its way, if it fails send back a
> > # stateless error to the requestor
> > if (t_relay())
> > {
> > xlog("L_INFO", "$ci|pass|$rd:$rp");
> > }
> > else
> > {
> > xlog("L_ERR", "$ci|end|unable to relay message");
> >
> > sl_reply_error();
> > }
> >
> >
> > }
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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> ------------------------------
>
> Message: 3
> Date: Mon, 22 Apr 2019 09:15:44 +0300
> From: Liviu Chircu <liviu at opensips.org>
> To: users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] Integrating with Asterisk in the same
> box
> Message-ID: <a2bce359-3139-6258-1375-c2c3beaf851c at opensips.org>
> Content-Type: text/plain; charset=utf-8; format=flowed
>
> > Hi all. I have an installation of opensips 2.4 with control panel, in
> > a Debian 8 server.
> > I would like to integrate my Opensips installation with an Asterisk
> > 16. For the time being it has to be in the same box. So it would
> > enhance with Incoming/outgoing Trunks, IVR, Voicemail, Conference,etc.
> > The system at the moment has this setup
> >
> https://www.powerpbx.org/content/opensips-v24-debian-v8-mariadb-apache-v1
> > and i have installed a simple asterisk (Chan_SIP) as simple as
> > possible, without anything fancy. As an example of the Asterisk config
> > i used the guide
> >
> https://computingforgeeks.com/how-to-install-asterisk-16-lts-on-ubuntu-18-04-16-04-debian-9/
> .
> > I have left the default 5060 port to Opensips and the port 5090 to
> > Asterisk.
> So far, so good. OpenSIPS in front, Asterisk in the back -- you're on
> the right path.
> > I am looking what other config do i have to do, so i could create
> > users in the Opensips control panel and created automatically in the
> > Asterisk's database to read from.
> > Also what other do i have to do to make them interact seamlessly, please?
> Here is when it gets dicy -- there is no such software available yet.
> In order to achieve this, one idea would be to fork the Control Panel
> and start hacking away at extending it in order to also work with
> Asterisk (additional GUI interactions? additional server-side PHP logic
> for the Asterisk DB handling?
>
>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 22 Apr 2019 09:23:30 +0300
> From: Liviu Chircu <liviu at opensips.org>
> To: users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] async() and config actions call stack
> Message-ID: <771806a2-70a4-f8b2-beb0-970475cea13a at opensips.org>
> Content-Type: text/plain; charset=utf-8; format=flowed
>
> Hi Vitalii,
>
> Indeed - we have thought about this alternative ever since we introduced
> "resume route". However, the amount of work required to make this
> happen is immense, and would retard the development of other,
> potentially much more useful features. It is our belief that, for the
> moment, having to break down the logic into "before" and "after"
> sections is not that much of a deal-breaker (I can personally attest to
> that!). Here are two tips to make your script more readable:
>
> * try breaking it down into multiple files, grouped by business logic
>
> * try to have a naming convention for the resume routes, for example:
> "resume_lnp_dip"
>
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
> On 09.04.2019 18:28, Vitalii Aleksandrov wrote:
> > Hi opensips team and community,
> >
> > Want to share one headache I have which might be converted into a
> > feature request. It's about async() implementation. I use it, like
> > probably many of us, for db operations and http requests and it's so
> > complicated to insert an async() call so some already written and
> > tested config. The requirement to set a "resume_route" and continue
> > request processing in it forces to break a script into "before async"
> > and "after async" parts. If you have only one async() call during
> > request processing it's more or less manageable and when it comes to
> > many async() operations it becomes a nightmare.
> >
> > I'm dreaming about the way to just "return" from a "resume_route"
> > and continue script execution from the instruction next to the place
> > were async() was called. Frankly speaking I didn't check how it's
> > implemented and config execution part of opensips is a blackbox for
> > me. So I assume there might be some architectural reasons and
> > obstacles which don't allow to make it this way.
> >
> > Haven't found anything related in 3.0 roadmap. I'm sure this would
> > be very beneficial for all opensips users.
> >
> > Feedback and comments are appreciated.
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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> ------------------------------
>
> End of Users Digest, Vol 129, Issue 20
> **************************************
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