- sr-users - kamailio.org

Kamailio WebRTC Call Not Working
by Ankit Jayswal 21 Mar '22

21 Mar '22

Hi, I have configured Kamailio to register a SIP user over websocket. SIP users are getting registered over webphone like JSSIP fine but not able to establish the call scenario below. >From Twilio Trunk - INVITE - KAMAILIO - ASTERISK - play hello world - Dial SIP user 101@Kamailio_IP - Rings 101 and Answer - Hangup. Here, 101 is registered with Kamailio IP, and I am sending a call from Twilio SIP trunk to Kamailio server IP. Asterisk is on port 5080 and Kamailio is on port 5090 both are on the same host 45.118.163.244. This above call scenario is NOT working with Webphones registered via JSSIP. I think the second call leg from Asterisk -> Kamailio is not getting set up properly and ACK is not getting confirmed here, call connects for some time and no voice transmits and then disconnects. The same call scenario is working with users registered at Softphone like Zoiper perfectly fine. No idea why it is not working with a webphone. I have attached kamailio.cfg, Kamailio log and Asterisk's pjsip logs here. So, please guide me on this as soon as possible. -- Thanks & Regards, *Ankit Jayswal* | Specialist - Software Development

1 0

pseudo var for From and To sip uri port
by Patrick Karton 20 Mar '22

20 Mar '22

Hello, is there a pseudo variable for setting only sip uri port for From and To Headers like $rp for R-URI ?

1 0

dialog module issues (BYE Request issued) when topos is enabled
by Patrick Karton 19 Mar '22

19 Mar '22

i am using dialog and topos module. but i have an issue i think with topos module. When topos module is disabled all the call flow works well but when i enable topos module i have this weird behaviour : after Caller sends BYE and complete the call , i receive later a BYE Request issued by the dialog module towards Caller and Callee. Does anyone already had that issue ? modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 1024) modparam("dialog", "rr_param", "oovdlg") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "db_update_period", 30) modparam("dialog", "wait_ack", 1) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "end_timeout", 180) modparam("dialog", "default_timeout", 3600) modparam("dialog", "noack_timeout", 90) modparam("dialog", "early_timeout", 300) modparam("dialog", "dlg_extra_hdrs", "X-OOV-Hint: Dlg-hit\r\n") modparam("db_text", "db_mode", 0) modparam("db_text", "emptystring", 0) modparam("db_text", "file_buffer_size", 8192) modparam("db_text", "max_result_rows", 1000) modparam("topos", "storage", "db") modparam("topos", "db_url", "text:///home/sbc/database") modparam("topos", "mask_callid", 0) modparam("topos", "sanity_checks", 0) modparam("topos", "contact_mode", 0)

1 0

DMQ version compatability
by Ben Kaufman 19 Mar '22

19 Mar '22

Daniel, In a recent thread you stated the following regarding DMQ compatibility between versions: "if you do dmq replication between kamailio systems running different major versions, then it is likely to get memory leaks due to replication of data and most probably cannot be fixed. This is because internal structures of modules (also dmq commands) can change, practically what an instance does is not ensured to happen on the other instance. Just for example, from my mind, htable got some changes during past releases, dmq also has significant enhancements by getting support for more transport protocols." Is it possible to get a bit more clarification on this, particularly as it relates to DMQ_USRLOC? I'm hoping to do rolling updates, but don't want to end up in a situation where I've caused myself more problems than I've solved. Ben Kaufman

3 2

help to configure topos module configuration
by Patrick Karton 18 Mar '22

18 Mar '22

Caller <--> (internal_IP) Kamailio (external_IP) <---> Callee I have this set up of kamailio with 2 interfaces. i use topos module for topology hiding and what i want to do is to send external_IP in Contact Header of request relayed to Callee and internal_IP in Contact Header of response relayed to Caller. i noticed that : when i use topos module with contact_mode to 0 all Record-Route and Caller Via Header are removed and thats great. but i dont find way to put 2 different ip addresses in Request and Response Contact Header? when i use contact_mode to 2 Record-Route and Caller Via Header are not anymore removed thats bad for topology hiding. And its seems we can not use $xavu(_tps_=>contact_host) parameter of module to put 2 different ip addresses in Request and Response Contact Header. What i want is : - remove all Record-Route and Caller Via Header for request like when contact_mode to 0 - and put 2 different ip addresses in Request and Response Contact Header is it possible to achieve it ? Thanks.

2 2

Upgrading from version 5.2 to 5.4 - any unpublished words of wisdom?
by Kevin Greene 18 Mar '22

18 Mar '22

Hi: I have a 5.2 version of Kamailio running and I was thinking to get it upgraded to the 5.4 branch. The 5.2 is working well but I have heard the 5.4 is a bit more stable. Thus the question becomes, is it worth the risk to upgrade and what should I look out for. Thanks. Kevin

2 1

Dialog module and DMQ
by Olle E. Johansson 18 Mar '22

18 Mar '22

Hi! The documentation on the DMQ support in the dialog module is a bit short, so I will ask here for clarifications: If I want to enable a rough limit on concurrent calls - will all servers share a $DLG_count ? Will get_profile_size() return the number of dialogs belonging to a profile on ALL servers? Thank you! /Olle

1 0

Kamailio as stateless failover proxy server and nat
by ppq＠tuta.io 17 Mar '22

17 Mar '22

Hello, I need some help in creating a stateless failover balancer for two asterisk servers using kamailio. I followed in the steps of an old project here: https://txlab.wordpress.com/tag/xlab1/ Kamailio is running in a multihomed host, with public ip facing the internet and private ip facing the asterisk servers. I have partial issues with nat traversal when the external device is also behind nat. All works when the device initates a call/dialog, however when the asterisk server tries to send an "OPTIONS" keep alive to the end device, Kamailio receives the packet from asterisk but tries to send it to the private IP address of the device instead of the public ip:port. Same goes for calls. My kamailio.cfg is below. I am missing something obvious here, but can't find it. Thanks === CUT==== #!KAMAILIO ################################################################ # # This is a front-end SIP proxy configuration for the xlab1 project # See for more details: https://txlab.wordpress.com/tag/xlab1/ # The configuration requires Kamailio version 3.3.x # # This work is licensed under a Creative Commons Attribution 3.0 License # Author: Stanislav Sinyagin <ssinaygin(a)k-open.com> # ################################################################ #!define WITH_DEBUG ####### Defined Values ######### #### define DBURL "text:///opt/sipfe/etc/kamailio_db" #!define DBURL "text:///etc/kamailio/kamailio_db" #!define BACKEND_NET4 10.10.10.0/24 #!define BACKEND_NET6 2a02:200:8:100::/64 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=2 log_stderror=yes #!else debug=1 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=2 auto_aliases=no #listen=udp:195.x.x.78:5060 # to avoid simple attacks, use unconventional port #port=5060 # this makes it listen on all ipv6 addresses auto_bind_ipv6=0 dns_try_ipv6=no # let kamailio choose the right source IP address mhomed=3 # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ## Hardening sip_warning=0 server_signature=no user_agent_header="User-Agent: Kamailio Failover Balancer" ####### Modules Section ######## ## NOTE: ## mpath="/opt/sipfe/lib64/kamailio/modules_k/:/opt/sipfe/lib64/kamailio/modules/" loadmodule "db_text.so" loadmodule "jsonrpcs.so" #loadmodule "mi_fifo.so" loadmodule "tm.so" loadmodule "kex.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" #loadmodule "mi_rpc.so" loadmodule "path.so" loadmodule "domain.so" loadmodule "dispatcher.so" loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") modparam("path", "use_received", 1) modparam("domain", "db_url", DBURL) modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") #modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## route { if (!sanity_check()) { exit; } if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); xlog("L_NOTICE", "$si $rm $ru -- too many hops\n"); exit; } if (af==INET) { force_rport(); if (src_ip != BACKEND_NET4) { # SIP request packet client->backend if( !loose_route() ) { if (!lookup_domain("$td", "dattr_")) { xlog("L_ERR", "$si $rm $ru -- domain \"$td\" is not " + "found in domain table\n"); drop(); } if (!defined $avp(dattr_routeset)) { xlog("L_ERR", "$si $rm $ru -- attribute \"routeset\" is " + "undefined for domain $td\n"); drop(); } if( !ds_select_dst(4000 + $avp(dattr_routeset), "1") ) { drop(); } } if (nat_uac_test("19")) { xlog("L_INFO", "Routing NAT in-dialog $rm from $fu to $du\n"); if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } } add_path_received(); #rtpproxy_manage("cwei"); rtpproxy_manage("cw"); record_route(); } else { # SIP request packet backend->client # Invites from backend contain Route field and it should be used # to reach the registered client loose_route(); #rtpproxy_manage("cwie"); rtpproxy_manage("cw"); record_route(); } } else { # IPv6: no NAT traversal, just pure routing if (src_ip != BACKEND_NET6) { # SIP request packet client->backend if( !loose_route() ) { if (!lookup_domain("$td", "dattr_")) { xlog("L_ERR", "$si $rm $ru -- domain \"$td\" is not " + "found in domain table\n"); drop(); } if (!defined $avp(dattr_routeset)) { xlog("L_ERR", "$si $rm $ru -- attribute \"routeset\" is " + "undefined for domain $td\n"); drop(); } if( !ds_select_dst(6000 + $avp(dattr_routeset), "1") ) { drop(); } } add_path_received(); record_route(); } else { # SIP request packet backend->client record_route(); } } forward(); } onreply_route { if (af==INET) { if(src_ip != BACKEND_NET4) { # SIP reply packet client->backend #rtpproxy_manage("cwie"); rtpproxy_manage("cw"); fix_nated_contact(); } else { # SIP reply packet backend->client #rtpproxy_manage("cwei"); rtpproxy_manage("cw"); } } return(1); } # Local Variables: # mode: sh # sh-indent-after-if: 0 # indent-tabs-mode: nil # tab-width: 4 # End: ====CUT=====

1 0

kamctl monitor : statistics not displayed ?
by Youssef Boujraf 17 Mar '22

17 Mar '22

Dear I am tying to monitor the kamailio with "kamctl monitor" and I got this error. -e [cycle #: 1; if constant make sure server lives] -e Kamailio Runtime Details: kamailio 5.5.4 (x86_64/linux) now: Wed Mar 16 12:44:43 2022 up_since: Wed Mar 16 12:39:51 2022 uptime: 292 -e Transaction Statistics: /usr/sbin/kamctl: 245: column: not found -e Stateless Server Statistics: /usr/sbin/kamctl: 259: column: not found -e UsrLoc Statistics: /usr/sbin/kamctl: 273: column: not found -e Core Statistics: /usr/sbin/kamctl: 287: column: not found -e Shared Memory Statistics: /usr/sbin/kamctl: 301: column: not foundAny idea Best Regards -- Youssef BOUJRAF

2 6

kamctl or kamcmd way to enable/disable sections of configs
by Joshua Young 17 Mar '22

17 Mar '22

#!define MYSECTION" route("1"); #!ifdef MYSECTION route("2"); #!endif I have declarations like this in my kamailio.cfg file, and was curious if there was a way to use kamctl or kamcmd to enable or disable the defined value in a live system. Or if there is another approach/mechanism I'm unaware of the "Serial Forking Based on Q Value" stuff and use it in other parts, but I am looking for a general way to manually manipulate routing on live system using cli commands if possible... Thanks, Joshua

3 2

Kamailio in docker container with TLS enabled using Let's Encrypt
by Giovanni Tommasini 17 Mar '22

17 Mar '22

Hi guys, starting from Fred article https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ we move step forward creating this small PoC to deploy Kamailio in docker container with TLS enabled using Let's Encrypt https://github.com/evoseed/kamailio-tls-letsencrypt What do you think? Any feedback? Any mistakes to correct or improvements to make? Thanks a lot Have a good evening Giovanni Tommasini *evoseed s.r.l.* phone: +39 340 11 28 800 mail: giovanni.tommasini(a)evoseed.io site: evoseed.io address: Via Lucrezio 13, Trieste Italy

2 2

sanity from uri check
by Juha Heinanen 16 Mar '22

16 Mar '22

I have in config: modparam("sanity", "uri_checks", 3) /* RURI, From */ and then config call sanity_check() on incoming requests: if (!sanity_check()) { xnotice("Request from <$var(src_ip)> has invalid syntax\n"); exit; }; When incoming request has: From: sipp <sip:@192.168.255.95:5062>;tag=1 no notice is printed to syslog. >From RFC 3261: SIP-URI = "sip:" [ userinfo ] hostport uri-parameters [ headers ] userinfo = ( user / telephone-subscriber ) [ ":" password ] "@" user = 1*( unreserved / escaped / user-unreserved ) My understanding of that is that there needs to be at least one character before @ sign. -- Juha

2 29

properly removing First Via header received
by Patrick Karton 16 Mar '22

16 Mar '22

i want to remove the First Via Header received from uac to kamailio. so in my script i use remove_hf("Via") but in the relayed request i see that ";received=x.x.x.x;rport=xxxx" is left without any header before it. i think internally kamailio adds a lump before i call the remove_hf function. how to do it properly ? Thanks

2 2

Forking for scripts?
by Matthias Urlichs 16 Mar '22

16 Mar '22

Hello, Is it possible to run a background Python script within Kamailio? I.e. call "fork_process" and then a non-terminating "python_exec()" from the child? -- -- regards -- -- Matthias Urlichs

2 2

Question about a future 5.6 release
by Stefan-Cristian Mititelu 16 Mar '22

16 Mar '22

Hi, I want to ask you if there is an approximate timeline for a kamailio 5.6 release? We are planning an upgrade on our systems and are weighing different options. Thank you Stefan

2 1

typo in documentation
by David Villasmil 15 Mar '22

15 Mar '22

Hello all, I was trying to implement db_text and the documentation states it should be something like modparam("usrloc|auth_db", "db_url", "db_text:///tmp/kamailiodb") but that doesn't work, it's modparam("usrloc|auth_db", "db_url", "text:///tmp/kamailiodb") so this line should probably be updated: https://github.com/kamailio/kamailio/blob/master/src/modules/db_text/doc/db… Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 2

load to htable from flat file
by David Villasmil 15 Mar '22

15 Mar '22

Hello guys, Is this possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 5

RTPEngine inject DTMF
by Bendersky 15 Mar '22

15 Mar '22

Hello Everyone, I am trying inject DTMF and getting error message. ERROR:rtpengine:rtpe_function_call: proxy replied with error: No SSRC context present for DTMF injection I have full call rtpengine debug which I can provide upon request. Rtpengine version rtpengine-9.5.4.1-9.fc32.x86_64 Volga629

1 0

clean an easy way to get sip uri of contact header
by Patrick Karton 15 Mar '22

15 Mar '22

hello, i want a generic way to get the sip uri in contact header. for example if Contact: Sipp <sip:6565@ip;ov=65;gid=45>;tid=700 i want a way to get both <sip:6565@ip;ov=65;gid=45> (without header param) and <sip:6565@ip;ov=65;gid=45>;tid=700 (with header param) Thanks.

3 3

limit the length of Via branch parameter
by Patrick Karton 15 Mar '22

15 Mar '22

Hello, i would like to know if there is a way or option to limit the length of the value of branch parameter in Via header ? Thanks.

2 3

Table : address - address_reload (exported MI function) - expect to call this function in kamailio.cfg to reload in cache the address
by Youssef Boujraf 14 Mar '22

14 Mar '22

Dear, I use the address table to ENABLE UAC doesn't registererd in subscriber table just for emergency call then after the BYE to I remove it from the address table. I 'd like to purge the UAC from address table BUT NEED to RELOAD in CACHE the address table. To use the "kamctl address relaod" works BUT I want to reload the address table from my Kamailio.cfg (route). I found the function "address_reload" ( https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528769) but I have an error when trying to start kamailio. Best regards, -- Youssef BOUJRAF | Managing Director Direct Line: +32 (0)2 880 33 30 Mobile: +32 (0)477 94 86 89 Dubai Mobile : +971 (0)55 9928871 Fax: +32 (0)2 318 82 21 Skype: y_boujraf Email: yboujraf(a)by-research.be Web : www.by-systems.be/ BY-SYSTEMS sprl| Avenue Charles Plisnier 13 | B-1070 Anderlecht | Belgium | TVA: BE 0819.832.617______________________________________________________________________________ This communication, issued by BY-SYSTEMS SPRL, is confidential and we do not waive confidentiality by mistransmission. If you have received it by mistake please notify the sender immediately by reply and then delete it permanently from your system without copying it, using it for any purposes or disclosing its contents to any other person. Thank you for your cooperation. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of BY-SYSTEMS SPRL. Emails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, lost or destroyed, or contain viruses. Anyone who communicates with us by email is taken to accept these risks.

2 1

WebSocket ping is not being sent
by Xuo Guoto 14 Mar '22

14 Mar '22

Hello list, I have kamailio 5.5.3 (x86_64/linux) configured as an SBC in front of asterisk to front end webrtc clients. Browser clients running JsSIP 3.7.0 connect to kamailio via webrtc and kamailio sends the traffic back to asterisk using SIP. rtpengine is configured to relay the RTP packets. I need to send websocket pings every 10 seconds to the browser clients. I have the following config in kamailio: modparam("websocket", "keepalive_mechanism", 1) modparam("websocket", "keepalive_timeout", 10) modparam("websocket", "keepalive_interval", 2) modparam("websocket", "keepalive_processes", 2) modparam("websocket", "verbose_list", 1) If my understanding of the documentation is correct, I do not need to do any additional configuration. But looking at the websocket traffic, I do not see any ping packets in websocket. I am also using sipdump module to capture all packets sent and received by kamailio and the ping packets are not visible there either. What could be wrong here? Do I need to do any additional config? Thanks for every help! X.

3 5

Any clever ways to strip down a user portion of an RURI?
by Brooks Bridges 12 Mar '22

12 Mar '22

Have a need to strip out various parameters from the user portion of the RURI and/or the To (npdi, etc). Does anyone have any clever methods for doing so that don't involve just brute forcing through the individual parameters? Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

3 4

Interconnect with Zoom
by Matthias Urlichs 11 Mar '22

11 Mar '22

Hi, has anybody managed to connect to Zoom with Kamailio, ideally using rtpengine? I'm getting error 500 when I call them … -- -- regards, -- -- Matthias Urlichs

3 4

Kamailio + RADCLI
by Marrold 11 Mar '22

11 Mar '22

Hi, We're currently testing Kamailio 5.5.3 on Debian 11 compiled with RADCLI and we're experiencing an issue with radius authentication: Feb 23 08:54:22 redacted /sbin/kamailio[817172]: WARNING: <script>: RADIUS auth failed for redacted (IP:redacted:5060) Feb 23 08:54:22 redacted /sbin/kamailio[817173]: radcli: rc_avpair_new: rc_avpair_new: no attribute 1/1 in dictionary Feb 23 08:54:22 redacted2 /sbin/kamailio[817173]: ERROR: auth_radius [sterman.c:204]: add_cisco_vsa(): unable to add Cisco-AVPair attribute I don't believe this is directly a Kamailio issue but enabling debug logging hasn't revealed any clues ( The log lines remain the same without additional info ) This was working previously on 5.5.X on Debian 9 Has anyone experienced this before or have any suggestions on where to start looking? Thanks

2 2

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