El lun., 22 abr. 2019 a las 1:24, <[email protected]> escribió:
> Send Users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > 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 <[email protected]> > To: [email protected] > Subject: Re: [OpenSIPS-Users] Fwd: Opensips SNGTC Module > Message-ID: <[email protected]> > 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 > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.opensips.org/pipermail/users/attachments/20190422/3e852bab/attachment-0001.html > > > > ------------------------------ > > Message: 2 > Date: Mon, 22 Apr 2019 09:10:54 +0300 > From: Liviu Chircu <[email protected]> > To: [email protected] > Subject: Re: [OpenSIPS-Users] WebRTC and mid_registrar issue > Message-ID: <[email protected]> > 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 > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.opensips.org/pipermail/users/attachments/20190422/37488446/attachment-0001.html > > > > ------------------------------ > > Message: 3 > Date: Mon, 22 Apr 2019 09:15:44 +0300 > From: Liviu Chircu <[email protected]> > To: [email protected] > Subject: Re: [OpenSIPS-Users] Integrating with Asterisk in the same > box > Message-ID: <[email protected]> > 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 <[email protected]> > To: [email protected] > Subject: Re: [OpenSIPS-Users] async() and config actions call stack > Message-ID: <[email protected]> > 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 > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------ > > End of Users Digest, Vol 129, Issue 20 > ************************************** >
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