sr-users March 2018

sr-users@lists.kamailio.org

101 participants
131 discussions

by Atux Atux

Hi. I am doing my first steps with Kamailio and i have installed it according to the http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour. i did create 3 users 1100,1101,1102 and then in windows PC i did install the jitsi application. Even though the accounts on Jitsi show as online on each PC i have audio issues. All PCs are in the same LAN -Device 1101 calls 1102 and on 1102 i cannot hear anything. 1102 talks and it cannot be heard -Device 1102 calls device 1101 and on 1101 PC i hear a ringing tone. Then i answer from 1101. 1102 speaks and it can be heard, not the other way around. any ideas, please?

6 years, 7 months

KEMI and SQL - best practices

by Igor Olhovskiy

Hi! Is there any suggestions, best practices of using SQL connections in KEMI? Main idea - not to open connection to database in every route (cause it’s really expensive operation in terms of time) and reuse existing, for ex, created in some kind of startup route? Or maybe use some else technique? Regards, Igor

6 years, 7 months

statistics (ASR and ACD) by destination/country

by Abdoul Osséni

Hello dear list, I am looking for a tool that will provide me call statistics (ASR and ACD) by destination/country with alerting by emails. I try to detect when a sip provider does not work well in order to route the traffic to another sip provider. Thank you. Abdoul OSSENI AfriCallShop

6 years, 7 months

KEMI basic logging

by Eugene Prokopiev

Hi, Tried just to reject any request with some log message: $ cat kamailio/Dockerfile FROM centos:centos7 RUN yum update -y RUN yum install -y wget RUN wget -O /etc/yum.repos.d/kamailio.repo http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… RUN yum install -y kamailio kamailio-lua VOLUME /etc/kamailio ENTRYPOINT ["kamailio", "-DD", "-E"] $ cat kamailio/config/kamailio.cfg listen=udp:0.0.0.0:5060 loadmodule "tm.so" loadmodule "sl.so" loadmodule "xlog.so" loadmodule "app_lua.so" modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") cfgengine "lua" $ cat kamailio/config/kamailio.lua function ksr_request_route() KSR.log("===== request - from kamailio lua script\n") KSR.sl.send_reply(503, "Server not configured") end Result is: kamailio_1 | 12(18) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 5 kamailio_1 | 0(1) ALERT: <core> [main.c:746]: handle_sigs(): child process 7 exited by a signal 11 kamailio_1 | 0(1) ALERT: <core> [main.c:749]: handle_sigs(): core was generated vms_kamailio_1 exited with code 1 But valid response '503 Server not configured' with commented KSR.log. So, how to log propertly? -- WBR, Eugene Prokopiev

6 years, 7 months

Re: [SR-Users] How to determine correct port to set in record-route header

by Pranathi Venkatayogi

Any help? From: Pranathi Venkatayogi Sent: Monday, January 23, 2017 2:35 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: How to determine correct port to set in record-route header Hi, I am using Kamailio behind NAT, unable to figure how to make it put “public ip” in Record-route header, I am manually inserting the hard-coded header myself as below. However now I am having trouble choosing the right port number in all scenarios. $var(dstPort) = 5061; if (dst_port==5060) { $var(dstPort) = 5060; } insert_hf("Record-Route: <sip:MY_PUBLICIP_ADDR:$var(dstPort);transport=tls;lr;nat=yes>\r\n", "Record-Route"); In one scenario I see the conflicting port numbers in “via” header vs “Record route”. The ACK is being set to port 5060 based on Record route header and is not being received by the callee. 2017-01-23 14:52:49.970233 [blink.exe 2780]: SENDING: Packet 11, +0:00:36.411867 10.0.27.108:58217 -(SIP over TLS)-> 172.31.211.31:5061 SIP/2.0 180 Ringing Via: SIP/2.0/TLS 63.149.103.72:5061;received=172.31.211.31;branch=z9hG4bK4799.7067bd2a48063748a4c353fa408eaefa.0 Via: SIP/2.0/UDP 10.0.16.52;rport=5060;branch=z9hG4bK4799.935a27c6ef5ad112225964cbe7c1be44.0;i=2 Via: SIP/2.0/TLS 10.11.200.12:51793;rport=51793;received=10.11.200.12;branch=z9hG4bKPj49f6aeca3f9b4155ab4c5304b544aa4d;alias Record-Route: <sip:63.149.103.72:5060;transport=tls;lr;nat=yes> Record-Route: <sip:10.0.16.52:5061;transport=tls;lr;nat=yes> Call-ID: 09ce10efa6a946bf9445ccc21857393e From: "cust1" <sip:cust1@devtranslation.sms-test.cyracom.com>;tag=95082caabecc42548c2fec5ccd29e5de To: <sip:spanish@translation.sms-test.cyracom.com>;tag=ec365cc8489b48a7bc5725d21b7d97a1 CSeq: 17422 INVITE Server: Blink 3.0.0 (Windows) Contact: <sip:20745891@10.0.27.108:58216;transport=tls> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Content-Length: 0 Questions: 1. How can I let in-built record-route function automatically choose “public ip” so I can get rid of my manual insertion altogether. 2. If no option, what is the right way to choose the port on which packet is being received, so it is same as what is on “VIA”. 3. Any other pointers to improve the logic here?

6 years, 7 months

kamailio it does not start after reboot

by Atux Atux

i am installing Kamailio using apt in a debian 8 machine by following this guide http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour i managed to create a few users and through Jitsi to talk to each other. once i reboot the machine, then the problems start and Kamailio cannot start. here are a few details from my system. -/etc/apt/sources.list the part for Kamailio #Kamailio deb http://deb.kamailio.org/kamailio51 jessie main deb-src http://deb.kamailio.org/kamailio51 jessie main -Version info Distributor ID: Debian Description: Debian GNU/Linux 8.10 (jessie) Release: 8.10 Codename: jessie -root@debian:~# systemctl start kamailio Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. root@debian:~# /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceJob for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. failed! Any ideas, please?

6 years, 7 months

180 and 183 SIP messages

by Ali Taher

Hello everyone, I'm working on freeswitch , where I have SIP calls coming from A-Party to my freeswitch and I'm routing it to B-Party. I'm facing an issue , when I'm getting back the 183 message from the supplier(B-Party) and forwarding it to the customer(A-Party); the customer is not hearing the RBT and asking to send 180 rather than 183 . (the supplier is only sending 183 without 180). Is it possible using kamailio to replace this 183 message with 180 message , or even add 180 message while keeping the same 183 message ? Regards, Ali Taher

6 years, 7 months

kamailio & vmware

by Jean Cérien

Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts. Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related. Anyone has advice on kamailio on a VM, when it only handles sip ? Rgds J.

6 years, 7 months

Kamailio fails to start because of jsonrpl

by Dmitry

Kamailio fails to start and I see the following errors: kamailio[2115]: ERROR: <core> [core/pvapi.c:1085]: pv_parse_spec2(): bad tr in pvar name "jsonrpl" kamailio[2115]: ERROR: <core> [core/pvapi.c:1111]: pv_parse_spec2(): invalid parsing in [$(jsonrpl(body){kz.json, result.NRSETS})] at (4) kamailio[2115]: CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kazoo/kamailio/dispatcher-role.cfg, line 243, colu...ult.NRSETS}) I installed kamailio-json-modules, Kamailio v5.1.2 installed from DEB packets from deb.kamailio.org What is the cause of the error?

6 years, 7 months

Kamailio to Asterisk Proxy with NAT

by Benjamin Marty

Hello, I have multiple Asterisk Servers in a private Network. They also have a Public IP via Destination NAT. Now I want to use a Kamailio Proxy in front of them. The Routing I want to do with the Kamailio Dispatcher module. My Question is now how the RTP Media Stream should/can flow. The clients are in different other networks. So P2P Media Stream isn't possible. Should I now run the RTP Stream Client - Asterisk or Client - Kamailio - Asterisk? Is there an Example for an such scenario? Thanks Benjamin

6 years, 7 months

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sr-users March 2018