sr-users April 2020

sr-users@lists.kamailio.org

70 participants
99 discussions

Topos_redis and loose_route
by Ernest Mavrel 15 Sep '20

15 Sep '20

Dear Community, Call scenario: Calling number 33825462354 Called number 44656646820 UAC (IP=1.1.1.1) => Kamailio (IP=2.2.2.2) => SIP proxy (IP=3.3.3.3) I stocked on strange issue with module TOPOS_REDIS and PRACK message (IP=2.2.2.2 kamailio version 5.2.3). Configuration with module TOPOS works, but because of a lot of calls we would like to use TOPOS_REDIS (avoid mysql). I already check this: https://lists.kamailio.org/pipermail/sr-users/2018-May/101641.html and I already have fixed version of module. In attach you can find traces (pcap file and kamailio log with debug=4) Your help will be greatly appreciated Kind Regards Ernest Mavrel

3 7

RTP port range in kuberenetes for kamailio
by Gaurav Bmotra 08 Jun '20

08 Jun '20

hi i have issue audio issue with kamailio in kubernetse cluster anybody , idea how to fix it in kubernetses thank you -- *Regards:* Gaurav Kumar

3 4

nathelper without usrloc
by Daniel W. Graham 18 May '20

18 May '20

I’ve been missing with this for a while and seem to be missing something. Any suggestions on what is missing here? Trying to use set_contact_alias() and handle_ruri_alias() from nathelper module and nat_keepalive from nat_traversal module, without registrar. I had register keepalive working, that has since broke. When register keepalive was working, I was able to place call in either direction but ACK and BYE was not being routed past kamailio. Registrations are forwarded to the PBX using add_path() and is working. Also not included below is the routing to the PBX, that is just setting $du and t_relay, and is also working. Topology is: UA1 -> NAT -> kamailio -> PBX -> UA2 Using default config file as the example, modified with above changes. I also removed RTP config as that is a non-issue. request_route { …… # FLAG MESSAGES FROM PBX setflag(FLT_PBX); route(NATDETECT); …… route[NATDETECT] { if (nat_uac_test("19")) { force_rport(); set_contact_alias(); nat_keepalive(); } return; } route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { route(DLGURI); } else if ( is_method("ACK") ) { route(NATMANAGE); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not here"); exit; } route[NATMANAGE] { if(isflagset(FLT_PBX)) { handle_ruri_alias(); } if(!isflagset(FLT_PBX)) { set_contact_alias(); } return; } route[DLGURI] { if(!isdsturiset()) { handle_ruri_alias(); } return; } branch_route[MANAGE_BRANCH] { route(NATMANAGE); } onreply_route[MANAGE_REPLY] { if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; -dan

4 19

imc module
by nicola defilippo 11 May '20

11 May '20

Hi, i'm a newbie about kamailio, i'd like create an imc chat. I load the module but i don't understand what's the sip uri to use to create the chat (i understand that there is a sip uri for the chat manager, but i don't know what it is). I' became a bit crazy. Thanks for help and sorry for the stupid question (if it's). N.

2 2

db_redis and PUA
by Carsten Bock 08 May '20

08 May '20

Hi, quick question: Did anyone ever attempt to use PUA with the new db_redis? If yes, could someone share the definition for this? Thanks, Carsten -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

3 2

Kamailio with dispatcher and asterisks real time
by ospos web 08 May '20

08 May '20

Hi all folks, Hello people, I'm new and I've been reading, so I have some but not much knowledge. I must integrate as work, two asterisk to the kamailio, a month ago I started with the real time guide [1] with only one kamailio, register, start the call, etc.. I am with the basics and it works. Now I must put two asterisk and using dispatcher, but although I read the documentation of the module.. use it and so then configure a list of file (with the asterisk where to dispatch) and not in the database.. MY HELP REQUEST QUESTION: How do I use dispatcher in load balancing mode, but taking in consideration my already working realtime-asterisk worling setup? THE PROBLEM: since with what I did it simply sends the call to both asterisk. i setup only 4 easy steps.. load module, then setup params (where i set the asterisk lists by file, and not db) and then added a "ds_select_dst(1, 4);" before the "route(RELAY);" line. please help! as i know.. the ds_select_dst(1, 4); prepare wicht asterisk will be choose to use and then when the routing RELAY happends no cares if in the realtime previously guide are a main asterisk? right? NOTE: I know it's a big world... but I can't keep reading the kamailio theory, i must goon forward due i have to lear later once are property working.. [1] https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

3 5

Releasing Kamailio v5.3.4
by Daniel-Constantin Mierla 06 May '20

06 May '20

Hello, based on the discussions from yesterday's online devel meeting, I am planning to release next week (likely on Wednesday, May 6) the next minor version out of branch 5.3, respectively 5.3.4. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Kamailio & rtpproxy with more than one external IP interfaces
by Leonid Fainshtein 30 Apr '20

30 Apr '20

Hi, My server has several external IP interfaces and one internal interface. For some calls I would like to handle RTP via Ext1<->Int interfaces and for other calls: via Ext2<->Int. In order to achieve this, I don't define the '-l' rtpproxy parameter, and depending on the call, I set the required IP address as a parameter of rtpproxy_manage(). I didn't succeed to get the desired result. On my testbed, I have the following interfaces: Ext1: 192.168.0.31 Ext2: 192.168.0.32 Int: 10.159.65.1 When I try to create Ext2<->Int (192.168.0.32<-> 10.159.65.1), the packets that accepted on the internal interface are sent out from the Ext1(192.168.0.31) instead of Ext2 (192.168.0.32). It looks like the decision about the source address is made based on the IP routing table information (?). Also, I would like to have an advertised address for at least one external interface. Is it achievable with just one instance of rtpproxy? Or, it is necessary to run two rtpproxies in the bridged mode wit configured advertize addresses? Thank you in advance, Leonid

3 7

Call from Asterisk - kamailio - Teams
by sip user 30 Apr '20

30 Apr '20

Hi... I have integrated Asterisk with Teams using Kamailio like SBC, thanks this list.. I can make call from Teams to Asterisk throught Kamailio, I have not Callerid from Teams but I'm working in this.. The problem that I have is call from Asterisk to Teams.. So, I have configurated a PSTN in Kamailio to my Asterisk. In Kamailio.cfg I have: * PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "IP ASERISK" desc "PSTN GW Address" pstn.gw_port = "PORT ASTERISK" desc "PSTN GW Port" In request_route, when I received an INVITE, I have: if (is_method("INVITE|SUBSCRIBE")) { if(src_ip==IP ASTERISK) { record_route_preset("centralita.arcanos.es:5061 ;transport=$ add_rr_param(";r2=on"); } else { route(PSTN); } } So, If I received an INVITE from a src_ip!=IP ASTERISK, I route to PSTN and call to my Asterisk(works, without callerid, but works). I don't know if there are any best way to do that.. If I received an INVITE from a src_ip==IP ASTERISK, I set record_route_preset, but how can I send the call to Teams? What is wrong? What are I doing wrong? Thanks

1 0

Time taken for tls_fix_domains_cfg initialization to complete is 8 seconds
by mahesh b 30 Apr '20

30 Apr '20

Hi All, I have 10 client profiles and 10 server profiles along with default client and server profile in my tls.cfg These are the numbers on time taken for tls_fix_domains_cfg initialization to complete based on the number of client and server profiles i have. Number of profiles Time diff (Sec) No of tcp child processes Start time End time 5 *3.99* 42 16:47:01.836 16:47:05.826 6 4.69 42 16:59:29.335 16:59:34.027 7 *5.347* 42 17:04:30.271 17:04:35.618 8 *6.10* 42 17:18:41.109 17:18:47.204 9 *6.614* 42 17:22:36.869 17:22:43.483 10 *7.379* 42 17:27:46.676 17:27:54.055 Kamailio 5.1.9 version is running on the below Linux Kernel version : 3.10.0-693.el7.x86_64 Centos version : CentOS Linux release 7.4.1708 (Core) CPU : 2 cores with model name : Intel(R) Core(TM) i7-6700 CPU @ 3.40GHz [root@miv5000 ~]# cat /proc/meminfo MemTotal: 3882076 kB MemFree: 811244 kB MemAvailable: 2320356 kB Openssl verison : OpenSSL 1.0.2k-fips 26 Jan 2017 Now my question is : Is it ok for the api to take "8 seconds" to complete initialization in case of having 10 client and 10 server profiles along with default profiles in tls.cfg ? The time units taken for initialization to complete should it be in the order of milliseconds range or in seconds range ? Are there any more parameters related to tls configuration like "crl " which would play a factor in time taken for the initialization to complete. Currently at my application level , timeout is happening as the initialization is taking more time when more number of profiles are enabled. Please share your inputs on whether it is justifiable with my setup cpu/memory, it is ok for tls_fix_domains_cfg to take 8 seconds for the initialization to complete. Regards, Mahesh.B.

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sr-users April 2020