sr-users April 2020

sr-users@lists.kamailio.org

70 participants
95 discussions

by Carsten Bock

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

4 years, 3 months

Kamailio with dispatcher and asterisks real time

by ospos web

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

4 years, 3 months

Releasing Kamailio v5.3.4

by Daniel-Constantin Mierla

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

4 years, 3 months

Kamailio & rtpproxy with more than one external IP interfaces

by Leonid Fainshtein

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

4 years, 3 months

Call from Asterisk - kamailio - Teams

by sip user

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

4 years, 3 months

Time taken for tls_fix_domains_cfg initialization to complete is 8 seconds

by mahesh b

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.

4 years, 3 months

Next online devel meeting - Wed, Apr 29, 2020, 14:00UTC

by Daniel-Constantin Mierla

Hello, in order to plan the roadmap to next major release, it is time to organize another online devel meeting, so the developers and community members can syncronize and plan what should be done till Kamailio v5.4 will be out. Because freenode.net service is now requiring user registration to join IRC channels on their servers, we are considering some alternatives that would allow guest participation, one of them being using a matrix chat room. More details about it will be posted soon. I created a wiki page for the event at: * https://www.kamailio.org/wiki/devel/irc-meetings/2020a Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, April 29, 2020, at 14:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

4 years, 3 months

NAT and RTP

by Pavithra M

Hi, I have installed kamailio as a ims server in openstack and enabled WITH_NAT and rtpproxy . I have installed pcscf, icscf, scscf and hss in four different ubuntu instances . server - kamailio 5.3.3. When i make a call using zoiper client from external network , call is becoming a missed call without giving ACK and BYE. Openstack - pcscf - 10.40.10.3 (floating ip - 10.45.4.22) icscf - 10.40.10.12 scscf - 10.40.10.5 hss - 10.40.10.31 zoiper1 - 10.0.2.15 zoiper2 - 10.0.2.15 kindly help me in this regard. when i installed without using NAT and rtpproxy, the call went well. Since currently i am using Opensatck , it is my prerequisite to use zoiper client externally .so for me it is mandate to enable NAT and rtpproxy. Help me please!!!

4 years, 3 months

change module parameter in routing logic

by Ali Taher

Hi, Is it possible to change a module parameter inside the routing logic ? For example I want to change sqlcon parameter of sqlops module based on the originating IP. Regards,

4 years, 3 months

DNS SRV issue

by Melek Oktay

Hi All, I have a user, who has a SRV records like below, srv:betaproxy.siptelekom.com 1- SRV betaproxy siptelekom 4 hrs 20 0 5060 beta-proxy-dc01.siptelekom.com (X.Y.87.150) 2- SRV betaproxy siptelekom 4 hrs 10 0 5060 beta-proxy-dc02.siptelekom.com (X.Y.87.160) And I want to send the call to him/her, but one of the potential IP address could be failed. Assume that, first record (X.Y.87.150) is failed when I'm trying to send the packet. Later on, timeout occurs and our Kamailio try to send the packet to second one (X.Y.87.160) after period ot time. My question: Next time when I want to send another call to the same user, how Kamailio could use second (X.Y.87.160) SRV instead of trying to first one (X.Y.87.150) ? What is the elegant way to implement this ? Best Regards, Melek

4 years, 3 months

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sr-users April 2020