sr-users

sr-users@lists.kamailio.org

42 participants
37519 discussions

allow_source_address_group in failure route
by Marrold 16 Feb '22

16 Feb '22

Hello, I'm using Kamailio as a proxy between several voice gateways and if the call fails for whatever reason (far end is busy etc) I am forwarding to an Asterisk box to play an announcement. Within the failure route I am using allow_source_address_group() to ascertain if the failure response was from a Voice Gateway or Asterisk itself, however it appears to return the group for the first branch when the reply is from the second branch to Asterisk. To handle the failure I am adding some headers, and setting a new destination indirectly with ds_select_dst before t_relay. I am using 5.5.3 on Debian 11, using KEMI / Python. Does anyone have any ideas? Thanks Matthew

2 1

Missing contact in 180 ringing causes wrong routing
by Xuo Guoto 15 Feb '22

15 Feb '22

Hello all, I have a kamailio ( kamailio 5.5.3 (x86_64/linux)) with asterisk (Asterisk 16.2.1~dfsg-2ubuntu1) with following architecture ┌─────────────┐ ┌─────────────┐ ┌─────────────────────┐ │ Asterisk ├───────┤ Kamailio │ 188.xxx.59.171 ───────── 103.xxx.xx.134 │ Linphone │ └─────────────┘ └─────────────┘ └─────────────────────┘ 10.10.1.13 10.10.1.9 In case the diagram is not displayed correctly its pasted at https://pastebin.com/b0UtE4Lj Registrations are handled by Kamailio and calls from linphone are routed to Asterisk. Similarly all calls originating in asterisk are routed over to Kamailio which in turn looks up AOR and routes the calls over to Linphone. Along with Linphone I also have a WebRTC endpoint at Kamailio. When calls are sent to WebRTC from asterisk, it gets correctly routed over to Asterisk, but if the client is Linphone the ACK of 200OK is not routed correctly to Linphone. I get this error in the logs: [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced tls:188.xxx.59.171:5061, to udp:188.xxx.59.171:5060) Digging into the difference between WebRTC and Linphone, I could see that in 1800 Ringing packet from WebRTC has a Contact headers while its missing in the packet from Linphone This is the 180 ringing from WebRTC: tag: rcv pid: 695615 process: 17 time: 1644860126.209155 date: Mon Feb 14 23:05:26 2022 proto: wss ipv4 srcip: 103.xxx.xx.134 srcport: 52508 dstip: 188.xxx.59.171 dstport: 443 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/WSS 188.xxx.59.171:443;branch=z9hG4bK730a.2cc7461b72f0c895e7cce5a0cedce4f7.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Content-Length: 0 While this is from Linphone tag: rcv pid: 695614 process: 16 time: 1644860097.970935 date: Mon Feb 14 23:04:57 2022 proto: tls ipv4 srcip: 103.xxx.xx.134 srcport: 25843 dstip: 188.xxx.59.171 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/TLS 188.xxx.59.171:5061;branch=z9hG4bKb5b1.d3231367383d07b1063098dcb35cc2b0.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPjd16f1379-1be7-448e-894a-c42d96f63146 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=cf7a7770-529d-4450-8bed-880382dce1dd To: <sip:user_jbjp@10.10.1.9>;tag=2ooYDia Call-ID: 8a025999-b096-4fe0-a623-9f6e6a69b8ae CSeq: 27838 INVITE User-Agent: Linphone Desktop/4.3.2 (Ubuntu 20.04.3 LTS, Qt 5.12.12) LinphoneCore/5.0.49 Supported: replaces, outbound, gruu Record-route: <sip:188.xxx.59.171:5061;transport=tls;r2=on;lr=on;did=3b9.b1b2> Record-route: <sip:10.10.1.9;r2=on;lr=on;did=3b9.b1b2> Content-Length: 0 When the 180 ringing is passed on to the Astrerisk it has the contact header modified with alias. tag: snd pid: 695615 process: 17 time: 1644860126.210189 date: Mon Feb 14 23:05:26 2022 proto: udp ipv4 srcip: 10.10.1.9 srcport: 5060 dstip: 10.10.1.13 dstport: 7000 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;alias=103.xxx.xx.134~52508~6;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Content-Length: 0 Now, I am not sure if this is the reason for routing the packets to wrong destination, but this is all I could figure out. Any clue to get this resolved will be much appreciated. My Kamailio config is pasted at https://pastebin.com/qyRS0usw with regards, X. Sent with ProtonMail Secure Email.

1 0

get kamailio sip processes id
by Patrick Karton 14 Feb '22

14 Feb '22

hello, as we are all aware kamailio create a lot of processes for many tasks. im using kamailio in kubernetes and i would like to monitor only sip processes with top command. i only need a way to get only kamailio sip processes id (udp receiver that handle sip request) . but i dont knwo how to get it. top -c -p id1,id2,id3,id4 Thanks

3 5

SEMS license with kamailio and rtpengine
by Patrick Karton 14 Feb '22

14 Feb '22

Hello im using kamailio and rtpengine to build a custom SBC. i was using freeswitch as my b2bua but i noticed a lot of issues when cps and simultaneous increased so i tried sems and i have better results. But i do not really understand sems license so i want to know if i can use it. i read from sems github "It is licensed under dual license terms, the GPL (v2+) and proprietary license. This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed. For a license to use SEMS under non-GPL terms, please contact FRAFOS GmbH at info(a)frafos.com" so my question is if i want to use sems as b2bua with kamailio and rtpengine and make profit of it do i need to purchase a license each time i want to sell it to a customer ? Thanks.

7 39

Kamailio server behind NAT
by Tihomir Culjaga 13 Feb '22

13 Feb '22

Hi all I'm wondering what is best practice to configure kamailio for a setup shown below. Carriers only support interconnection over a private network so I need to terminate every separate network on kamailio machine (and on RTPEngine) . That will turn very quickly into an administration nightmare as we add more and more carriers. My plan is to terminate carrier private networks on a router and NAT the traffic to kamailio where it leads me to a kamailio behind NAT setup. Well, in theory set_advertised_address and set_advertised_port per message can do the trick... at least I hope. Also, some carriers require a SIP registration to activate SIP trunks. I do that using UAC module (remote registration) but here again I got a problem with contact address as you can set only one contact address (modparam("uac", "reg_contact_addr", "10.1.1.11:5060")). /--RTP 10000-19999-- RTPEngine1 -----RTP------+-- IPPBX1 carrier1 --- NAT ---| | | | | | carrier2 --- NAT ---+-----SIP 5060------ kamailio ----SIP---- Dispatcher | | | carrier3 --- NAT ---| | | \--RTP 20000-29999-- RTPEngine2 -----RTP------+-- IPPBX2 So bottom line, I'm wondering what would be the best practice for such a scenario ? That would spare me from hitting many pitfalls. Would really appreciate a hint. Regards, Tihomir.

3 3

Vo5G with open5gs
by Hornbuckle, Mark A 11 Feb '22

11 Feb '22

Hello, I have been trying to find documentation on Kamailio IMS and Vo5G support for open5gs. I wanted to ask you directly if Kamailio IMS does support Vo5G with open5gs? Thank you

1 0

resending same message from UE after receiving RP-ACK
by Eugene B 11 Feb '22

11 Feb '22

Hi, I trying to use smsops module as SMSC. When I sending message from UE I see re-sending same message after each RP-ACK(~10 attempt). What could be the problem? Thanks! pcap in attach. kamailio 5.3.2 -- *Yours faithfully,* *Eugene B.*

2 1

kamilio open5gs 5G SA compatible?
by Sandland, Seth P 11 Feb '22

11 Feb '22

Were looking to test a kamilio on Open5gs 5g stand alone ( non lte, non NSA) can you conform or provide documents or install instructions on how this is accomplished. I can not find much of any details if Kamilio support Open5Gs 5g SA and what features it does support on a 5g SA core. Seth Sandland Wireless Communications IT Engineer Pacific Northwest National Laboratory Tel: (509) 375-6842 Email: seth.sandland(a)pnnl.gov<mailto:seth.sandland@pnnl.gov> www.pnnl.gov<http://www.pnnl.gov/> [A close up of a logo Description automatically generated]

1 0

Publish XML and xpath
by Igor Olhovskiy 10 Feb '22

10 Feb '22

Hello, I'm trying to get state of endpoint from PUBLISH message in a most simple way using xmlops module Trying to make this $xml(body=>doc) = $rb; log("L_ERR", "$rU is now in state $xml(body=>xpath:/dialog-info/dialog/state)\n"); XML in PUBLISH itself <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:88881@dev.company"> <dialog id="v6qmEYf5yI" direction="initiator"> <state>confirmed</state> </dialog> </dialog-info> Actual text in log is ... <script>: 88881 is now in state Means it's empty. xmlops module is loaded with no parameters. Tried also string $xml(body=>xpath:/dialog-info/dialog/state/text()) -> same result What am I missing here? kamailio 5.4.6 -- Thanks in advance, Igor

2 5

KEMI KSR.async Python3
by Ahmed Moubayed 10 Feb '22

10 Feb '22

Hi, I'm trying to use the async module from a python3 Kemi script but it conflicts with the keyword async. Any suggestions on how it can be used? Is there an alternative to KSR.async.route()? Thanks, Ahmed

1 0

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users