sr-users August 2017

sr-users@lists.kamailio.org

89 participants
144 discussions

by Alex Balashov

I have gathered that RTPEngine has recently, or perhaps some time ago, evolved a recording feature set: https://kamailio.org/docs/modules/5.0.x/modules/rtpengine.html#rtpengine.f.… Does anyone have any experience using it with Kamailio? How does it work? Any gotchas or pitfalls? It looks like recording is done via a separate daemon specifically for that purpose. Does it emit directly in a playable format, or more or less just dump the raw, RTP-encapsulated frames? -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

6 years, 9 months

DNS Geo Location Kamailio servers, registration and routing

by Sadeesh Radhakrishnan

We have kamailio servers that do DNS-Geo-Location on different locations, using same domain name. So the users from a country will be preferred to his nearby server by DNS routing. Similarly users from different countries will be registering to their nearby kamailio sip servers. All the servers share a common user location table on database. Now the users are getting registered fine, with no issue. We want the users at one location should be able to call user@other-location using the user location contact information. When we tried, the INVITE is sent on common domain name and that reverse resolves dns to same server and gets a 404 not found. I need some suggestion / ideas in order to make it happen. Thanks

6 years, 9 months

set digest mime

by wsotest.512

6 years, 9 months

Inconsistency in sems v1.6 installation: sems.conf is missing

by leslie mekiyess

For some reason, the installation creates the etc directory under the /usr/local/etc/sems/ : make[2]: Entering directory `/usr/src/sems-1.6.0/apps/conf_auth' mkdir -p **/usr/local/etc/sems/**etc**/** installing conf_auth.conf make[2]: Leaving directory `/usr/src/sems-1.6.0/apps/conf_auth' Then at the make install ends the following is printed: *** install complete. Run SEMS with *** *** /usr/local/sbin/sems -f /**usr/local/etc/sems/sems.conf** At bottom libe, while following the spain document instructions, I figured out that the sems.conf wasn;t created at the /usr/local/etc/sems/ path. I can create a new one or even copy this one found : /usr/src/sems-1.6.0/apps/mobile_push/load_test/sems_cfg/core/etc/sems.conf but I am afraid of further files/directory inconsistencies. Any assistance is very appreciated.

6 years, 9 months

acc billsec

by Aidar Kamalov

Hello, is there any way to log in cdr duration and billsec? -- Aydar A. Kamalov

6 years, 9 months

UAC RPC Commands problem

by Volkan Oransoy

Hi, I use Kamailio 5.0. I try to run RPC commands for UAC module but some of them doesn’t work like documented. kamcmd uac.reg_dump kamcmd uac.reg_reload These commands work as expected. But commands with extra arguments give error. kamcmd> uac.reg_refresh 908508850000 error: 400 - Invalid Parameters kamcmd> uac.reg_info l_uuid 908508850000 error: 404 - Record not found Do I miss something? Thanks, /Volkan

6 years, 9 months

Carrierroute, fallback, reinvites

by yu＠yu-boot.ru

Hello. Trying to implement fallback route for carrierroute and got sick to make reinvites working correctly. Until fallback route triggers, everything is OK. Anyone has working config for that purposes (NAT traversal, reinvites, carrierroute ), please? And abstract question - is it generally a good idea to implement a simple softswitch (not PBX like Asterisk) like Yate or Freeswitch with Kami? Now I just want to learn Kami and learn SIP deeper in general.

6 years, 9 months

Route calls from Kamailio to Asterisk behind NAT

by vs＠cell.tel

6 years, 9 months

Re: [SR-Users] [OpenSIPS-Users] ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC

by Abdul Basit

nice presentation!!! -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On 10 August 2017 at 23:12, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > *Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for > SIP and WebRTC* > > Scaling your FreeSWITCH platform to serve a growing user base is a > critical challenge. We'll go through the best techniques, practices, and > implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, > Chatting, Presence and Instant Messaging > Slides can also be downloaded from http://opentelecom.it/cluecon2017.pdf > > Enjoy ClueCon! > > -giovanni > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >

6 years, 9 months

Dispatcher call id hash for server side invite

by Ryan Wagoner

For a call invite from a phone to an load balanced asterisk server farm I can use ds_select_dst with hash over callid for the algorithm. What I don't understand is what happens for a server side invite. Say user A calls user B. The server will send and invite to user B's device. User B's device will reply and ds_select_dst won't have the call id hash so it will choose a random server, which might not be the server sending the invite. How do I mark these server side invites so the call hash is known by kamailio? Or am I thinking about this the wrong way? Thanks, Ryan

6 years, 9 months

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sr-users August 2017