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37 participants
37511 discussions

UAC RPC Commands problem
by Volkan Oransoy 14 Aug '17

14 Aug '17

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

2 3

Carrierroute, fallback, reinvites
by yu＠yu-boot.ru 12 Aug '17

12 Aug '17

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.

2 1

Route calls from Kamailio to Asterisk behind NAT
by vs＠cell.tel 12 Aug '17

12 Aug '17

3 5

Re: [SR-Users] [OpenSIPS-Users] ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Abdul Basit 11 Aug '17

11 Aug '17

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 > >

1 0

Dispatcher call id hash for server side invite
by Ryan Wagoner 11 Aug '17

11 Aug '17

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

3 6

Issue in Dialplan Module
by Logeshwaran G 10 Aug '17

10 Aug '17

Hi, I am facing an Problem in Dialplan Module. Scenario: ======== I am Dialling 00091xxxxxxxxxx number, I need to remove only the prefix 000, then the call placement will be 91xxxxxxxxxx. My dialplan rule: ============== MariaDB [kamailio]> select * from dialplan; +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ But the prefix 000 is not replaced. Please Guide me to Resolve this issue. Do I need to change somet rules?? Additional info: ============= But, With additional to the above dialplan data rule I used the below rule. +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | 111 +----+------+----+----------+-----------+-----------+-----------+----------+-------+ The above rule replacing only the 000, If we Dial 000xxxx its omitting, While Dialing 000 its replacing that with 111. Thanks & Regards, Logeshwaran

4 23

BUG: invalid branch
by Yasin CANER 10 Aug '17

10 Aug '17

2 1

Kamailio: how to route RTP media directly to endpoint
by wsotest.512 10 Aug '17

10 Aug '17

Hi all, We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that: UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider. So we decided to try route all rtp traffic between users directly. UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> --> --> UserB Is it possible at all? Maybe someone already did it . -- BR, Alex

3 2

IMPORTANT: acc module - SQL define removed and diameter code relocated
by Daniel-Constantin Mierla 09 Aug '17

09 Aug '17

Hello, I just pushed a bit of clean up to acc module, therefore be aware of following changes: 1) the define conditions on SQL_ACC were removed -- this was enabled for more than 10 years and only made the code look complex and hard to follow up its logic. 2) the code related to DIAMETER accounting was relocated to acc_diameter (new) module. It was a consistent size of code that was not enabled for sooo... long. It is now a dedicated module, similar to acc_radius. The diameter accounting code, even a new module now, is in the same stage, compiling but not tested, in pair with auth_diameter module, it may work, but very likely not. In summary, what's important for those using the acc module -- it offers the same functionality as it was enabled by default in the past 10 years or more, only the unused code was relocated. It offers the functionality of writing accounting records to syslog and sql databases. The acc module is now slimmer, only with the code that it needs, therefore easier to maintain and enhance for the future. Hopefully, there was no side effect with this update -- anyhow, if you find any issue, just open a bug report on github project. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Kamailio: how to route RTP media directly to endpoint
by Aleksey Hodin 09 Aug '17

09 Aug '17

1 0

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