- sr-users - kamailio.org

SDPOPs questions
by Gerry | Rigatta 23 Mar '21

23 Mar '21

Hi, The behaviour of the sdpops module is not fully clear to me. Lets say in the INVITE contains in the m= line the payloads 0 99 but has no a= line for codec 0 / PCMU, Would the sdp_keep_codecs_by_name(“PCMU”) keep PCMU ? What about the case that the INVITE contains in the m=line only the payload 99 but shows an additional a=line for codec 0 / PCMU. Would the sdp_keep_codecs_by_name(“PCMU”) keep PCMU ? If yes, would the extra mtype parameter prevent that? e.g. sdp_keep_codecs_by_name(“PCMU”, mtype) What happens when several m= lines are present? Does sdp_keep_codecs_by_name consider all m=lines ? Are the treated separately? What happens to payloads which are not in the hard-coded list in sdpops_data.c ? Can they be used with the sdpops functions? Thanks for your hints. Best Gerry codecsmap_t sdpops_codecsmap_table[] = { { {"PCMU", 4}, {"0", 1} }, { {"GSM", 3}, {"3", 1} }, { {"G723", 4}, {"4", 1} }, { {"DVI4", 4}, {"5,6,16,17", 9} }, { {"LPC", 3}, {"7", 1} }, { {"PCMA", 4}, {"8", 1} }, { {"G722", 4}, {"9", 1} }, { {"L16", 3}, {"10,11", 5} }, { {"QCELP", 5}, {"12", 2} }, { {"CN", 2}, {"13", 5} }, { {"MPA", 3}, {"14", 2} }, { {"G728", 4}, {"15", 2} }, { {"G729", 4}, {"18", 2} }, { {0, 0}, {0, 0} } };

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Dynamic CSEQ Updates
by Tyler Moore 23 Mar '21

23 Mar '21

Hey kamailio team, When are the callbacks in the dialog module for updating $dlg_var(cseq_diff) called? I am trying to dynamically enable / disable cseq updates (some specific carriers need it and some don't) by setting $dlg_var(cseq_diff)=$null prior to sending the request but it seems like the callback happens after relaying. For example I tried various locations: - prior to t_relay() - in onreply_route[] - in failure_route[] - in event_route[tm:local-request] - in event_route[tm:local-response] I also tried setting modparam("dialog", "initial_cbs_inscript", 1) and calling dlg_manage() prior to routing and those didn't seem to be the right callbacks either. Either the callback that updates $dlg_var(cseq_diff) occurs after t_relay() / forward() is called or setting $dlg_var(cseq_diff)=$null has no effect. Is dynamically turning CSEQ updates on / off possible by overwriting $dlg_var(cseq_diff)? Or would this require new logic in the dialog module to be supported? Regards, *Tyler Moore* <https://devopsec.net/> Full Stack Software Engineer <http://www.google.com/search?q=full+stack+software+engineer+job+description> Flyball Labs <http://goflyball.com/> Office: 888-907-2085, ext: 34 <8889072085;ext=34> Cell: 248-909-2769 <2489092769> Email: tmoore(a)goflyball.com [image: FLYBALL TECHNICAL SOLUTIONS, LLC]

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Info: Freezing development for Kamailio v5.5.x series - March 31, 2021
by Daniel-Constantin Mierla 22 Mar '21

22 Mar '21

Hello, this is the notification to let everybody know that the development of Kamailio v5.5.x series will be frozen at the end of March 31, 2021. Should anyone plan to add new features/modules and be part of v5.5.x, then they must be committed to git master branch or a pull request created by that date. Afterwards, the new features will have to be hold for a few weeks till branch 5.5 is created and development branch unfrozen, and they will be part of the next future release v5.6.x (or other version number to be decided later). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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JANSSON Delete from JSON
by Jeremy McNamara 22 Mar '21

22 Mar '21

Hi Folks - I am using JANSSON to store JSON data in an htable. How does one remove an entire element from a JSON? I do not see a specific function mentioning clear or remove exposed in the Kamailio module, but do see various clear/remove methods in the underlying C library. Below does set the value of key foo to null. I would like to remove both the key and value of foo. jansson_set("null", "$var(foo)", "$var(json)", "$var(result)"); Running Kamailio v5.4 Debian 10 -Jeremy

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Kamailio + rtpengine + Freeswitch on public/private network
by Eric Ravelo 20 Mar '21

20 Mar '21

Dear Kamailians, I have Kamailio+rtpengine as a dispatcher in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network (in an AWS VPC). Kamailio receives calls on the Public Internet IP address and port, and dispatches to FreeSWITCH on the private network. All calls are coming from a SIP Trunk provider, from the Internet so no NAT on this side. My Hello World scenario is: FreeSWITCH answers the calls, plays a media file and disconnects. Problems: 1) In the INVITE relayed from Kamailio to FreeSWITCH, I can see a Record-Route with the public IP of Kamailio. Hence, the BYE sent by FreeSWITCH is sent to this public IP, rather than going back to the Kamailio path, and the call never gets completed from the caller's point of view. I tried adding record_route() in request_route{} but nothing changes. 2) The SDP part of all the messages coming back and forth from Kamailio and FreeSWITCH all contain the correct private IPs, but still the media file can't be heard on the caller's side. Is RTPengine correctly bridging the RTPs? Kamailio configuration is here: https://gist.github.com/eravelo/b3f026723e27d42354804ef942b34a04 Rtpengine configuration file has: [rtpengine] table = 0 interface = internal/10.20.1.158;external/10.20.1.158!KAMAILIO_PUBLIC_IP listen-ng = 127.0.0.1:2223 timeout = 60 silent-timeout = 3600 tos = 184 port-min = 16384 port-max = 32768 Your help is appreciated. Thank you. Eric R.

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received parameter in route URI not used despite path use_received set to 1
by Easy Rider 19 Mar '21

19 Mar '21

Hi I'd like to make sure I've understood correctly how use_received in path module is supposed to work, because I don't get what I expected to get after reading the documentation I've got a little set up with a UAC behind a NAT registering to a kamailio registrar through a kamailio edge proxy (he UAC is a simple sipp script simulating a UAC behind a NAT) path is enabled on both EP and registrar, and use_received is set to 1 on EP the path, contact and via headers in the REGISTER going from the EP to the registrar are : Via: SIP/2.0/UDP 192.168.26.129;branch=z9hG4bK20da.9f4ab6ad25eae6acdd4bd4d655b320ce.0 Via: SIP/2.0/UDP 10.0.0.5:6060 ;received=192.168.26.1;rport=23456;branch=z9hG4bK-105-1-1 Contact: <sip:1000@10.0.0.5:6060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS" Path: <sip:inside@192.168.26.129;lr;received=sip:192.168.26.1:23456> where 192.168.26.129 is obviously the edge proxy then I place a call to the UAC through registrar the EP receiveds the following INVITE : INVITE sip:1000@10.0.0.5:6060 SIP/2.0 Record-Route: <sip:192.168.26.142;lr> Via: SIP/2.0/UDP 192.168.26.142;branch=z9hG4bKdf25.142eebc72e940d07ad983a858c67f64e.0 Route: <sip:inside@192.168.26.129;lr;received=sip:192.168.26.1:23456> Via: SIP/2.0/UDP 192.168.26.1:34567 ;received=192.168.26.1;rport=34567;branch=z9hG4bK-107-1-1 From: "1001" <sip:1001@my-fake-domain.com>;tag=107-1 To: <sip:1000@my-fake-domain.com;user=phone> CSeq: 1 INVITE Call-ID: 1-107(a)192.168.26.1 Contact: <sip:1001@192.168.26.1:34567> ... The Route header is exactly what is expected, but the edge proxy still sends the request to the request-URI address (10.0.0.5:6060) instead of sending the request to the received address set in the first Route header The path module documentation states : 3.1. use_received (int) If set to 1, the “received” parameter of the first Route URI is evaluated and used as destination-URI if present. kamailio configuration on the EP is almost the default one (just added path support, and modified the REGISTRAR route to forward the request instead on saving the contact, so there is not even a mid registrar on the EP tried with kamailio 5.3.7 and 5.4.4 thanks

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Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 18 Mar '21

18 Mar '21

I have one Kamailio that makes an outbound register on two Asterisk. I have a dispatcher with that same two Asterisk destinations, on INVITE message I call replace_from("", ) in order to call a internal extension and on failure route I receive 401 and call the process of authorization is OK. The problem is that if the first Asterisk fail and I get 500 on failure route I can not call replace_from() because is not valid on the block. Is there any way to make this type of authorization process after receiving a 500 on failure route of dispatcher ? Below my cfg file. Thanks in advance. Antonio Couto #!KAMAILIO#!ifdef WITH_DEBUG#!define DBGLEVEL 3#!else#!define DBGLEVEL 2#!endif # - flags# FLT_ - per transaction (message) flags# FLB_ - per branch flags#!define FLT_ACC 1#!define FLT_ACCMISSED 2#!define FLT_ACCFAILED 3#!define FLT_NATS 5 #!define FLB_NATB 6#!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0log_prefix="{$mt $hdr(CSeq) $ci} " children=8 auto_aliases=noenable_tls=noenable_sctp=no tcp_accept_no_cl=yes # SIP port listen=udp:192.168.0.201:5060# xhttp port listen=tcp:127.0.0.1:8181 loadmodule "xhttp.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "htable.so" loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 1) # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params -----# default retransmission timeout: 2sec modparam("tm", "fr_timer", 1500)# default invite retransmission timeout after 1xx: 40sec modparam("tm", "fr_inv_timer", 40000) # ----- rr params -----# set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) # account triggers (flags) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("db_text", "db_mode", 1) modparam("db_text", "emptystring", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "track_cseq_updates", 1) modparam("uac", "restore_dlg", 1) modparam("uac", "reg_db_url", "text:///etc/kamailio/dbtext") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_contact_addr", "192.168.0.201:5060") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations if (is_method("INVITE")) { route(DISPATCH); } else { sl_send_reply("501","Not Implemented"); exit; } } # Dispatch requests route[DISPATCH] { if ($si == "192.168.0.44" and $sp == 5060) { xlog("L_ALERT","from switch $si:$sp)\n"); $var(groupid) = "2"; } else{ xlog("L_ALERT","from others $si:$sp)\n"); $var(groupid) = "1"; } if(!ds_select_dst($var(groupid), "11")) { send_reply("503", "Service Unavailable"); exit; } if ($du =~ "192.168.0.210") { xlog("L_ALERT", "SCRIPT: uac_replace_from\n"); uac_replace_from("", "sip:4000@192.168.0.201:5060"); } xlog("L_ALERT", "SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { $var(ts) = t_get_status_code(); xlog("L_ALERT","ALERT: failure_route[RTF_DISPATCH]] from $si:$rp status:$var(ts)\n"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = "4000"; $avp(apass) = "1234"; $avp(arealm) = "asterisk"; if (uac_auth()) { route(RELAY); } exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if (ds_next_dst()) { xlog("L_ALERT", "retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("500","Server Internal Error"); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # jsonrpc dispatch event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } }

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[Question] About softphone for VoLTE
by shigeru ishida 18 Mar '21

18 Mar '21

Hi all, I have one question. What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I would like to verify VoLTE of private LTE with Open5GS and Kamailio. I'm sorry to ask you a question without much understanding of Kamailio, VoIP and VoLTE. Thank you very much in advance. Best regards, --Shigeru

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Server requirements estimation
by Marat Gareev 17 Mar '21

17 Mar '21

Hello! I'm configuring Kamailio proxy and trying to find instructions or advises for choosing the following options: *Host specific* - Number of cores - RAM size *Kamailio specific* - Number of children to fork for the UDP interfaces - Shared memory size The proxy will not connect to the database, only route INVITEs using the dispatcher. The expected load is 70K requests per hour. Are there any formulas for calculating the parameters?

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sip_trace() in branch_route, kamailio 5.1
by Daniel Donoghue 17 Mar '21

17 Mar '21

Hello, Using kamailio 5.1, I am trying to use sip_trace(<internal-sip-uri>) from a branch route to send a copy of the current sip message to another location. Sending the message works fine but I note that the message being sent is the original incoming request and does not contain what was changed for the particular branch. Is there any way to get sip_trace() to send the content of the message for that specific branch? With every blessing, — Daniel Donoghue

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Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 17 Mar '21

17 Mar '21

I have one Kamailio that makes an outbound register on two Asterisk. I have a dispatcher with that same two Asterisk destinations, on INVITE message I call replace_from("", ) in order to call a internal extension and on failure route I receive 401 and call the process of authorization is OK. The problem is that if the first Asterisk fail and I get 500 on failure route I can not call replace_from() because is not valid on the block. Is there any way to make this type of authorization process after receiving a 500 on failure route of dispatcher ? Below my cfg file. Thanks in advance. Antonio Couto #!KAMAILIO#!ifdef WITH_DEBUG#!define DBGLEVEL 3#!else#!define DBGLEVEL 2#!endif # - flags# FLT_ - per transaction (message) flags# FLB_ - per branch flags#!define FLT_ACC 1#!define FLT_ACCMISSED 2#!define FLT_ACCFAILED 3#!define FLT_NATS 5 #!define FLB_NATB 6#!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0log_prefix="{$mt $hdr(CSeq) $ci} " children=8 auto_aliases=noenable_tls=noenable_sctp=no tcp_accept_no_cl=yes # SIP port listen=udp:192.168.0.201:5060# xhttp port listen=tcp:127.0.0.1:8181 loadmodule "xhttp.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "htable.so" loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 1) # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params -----# default retransmission timeout: 2sec modparam("tm", "fr_timer", 1500)# default invite retransmission timeout after 1xx: 40sec modparam("tm", "fr_inv_timer", 40000) # ----- rr params -----# set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) # account triggers (flags) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("db_text", "db_mode", 1) modparam("db_text", "emptystring", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "track_cseq_updates", 1) modparam("uac", "restore_dlg", 1) modparam("uac", "reg_db_url", "text:///etc/kamailio/dbtext") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_contact_addr", "192.168.0.201:5060") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations if (is_method("INVITE")) { route(DISPATCH); } else { sl_send_reply("501","Not Implemented"); exit; } } # Dispatch requests route[DISPATCH] { if ($si == "192.168.0.44" and $sp == 5060) { xlog("L_ALERT","from switch $si:$sp)\n"); $var(groupid) = "2"; } else{ xlog("L_ALERT","from others $si:$sp)\n"); $var(groupid) = "1"; } if(!ds_select_dst($var(groupid), "11")) { send_reply("503", "Service Unavailable"); exit; } if ($du =~ "192.168.0.210") { xlog("L_ALERT", "SCRIPT: uac_replace_from\n"); uac_replace_from("", "sip:4000@192.168.0.201:5060"); } xlog("L_ALERT", "SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { $var(ts) = t_get_status_code(); xlog("L_ALERT","ALERT: failure_route[RTF_DISPATCH]] from $si:$rp status:$var(ts)\n"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = "4000"; $avp(apass) = "1234"; $avp(arealm) = "asterisk"; if (uac_auth()) { route(RELAY); } exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if (ds_next_dst()) { xlog("L_ALERT", "retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("500","Server Internal Error"); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # jsonrpc dispatch event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } }

2 1

kamcmd dlg.end_dlg not working in shell script
by Andrew Chen 16 Mar '21

16 Mar '21

Hi all, I have a script that cleans up hanging dialog and for some reason, executing 'kamcmd dlg.end_dlg' does not seem to work. Below I log events of stuck calls and as you can see it just keeps logging the same call until I manually run the kamcmd either in bash shell prompt or in kamcmd cli: 2021-03-08T19:00:04.347807+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:00:04 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:01:02.531822+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:01:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:02:02.689823+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:02:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:03:02.044960+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:03:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:04:02.436588+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:04:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:05:02.685572+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:05:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:06:02.775561+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:06:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:07:01.867021+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:07:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:08:01.888512+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:08:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:09:02.146887+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:09:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:10:02.649627+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:10:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:11:01.904948+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:11:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:12:01.932267+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:12:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:13:02.428128+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:13:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:14:02.617683+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:14:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:15:01.863099+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:15:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:16:01.650299+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:16:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:17:02.338576+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:17:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:18:01.743163+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:18:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:19:02.257369+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:19:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:20:01.624170+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:20:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:21:02.287343+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:21:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul Thoughts? Thanks! -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

5 15

dmq support for all transports - udp, tcp, tls, sctp
by Daniel-Constantin Mierla 16 Mar '21

16 Mar '21

Hello, (cross-posting - notification for developers, but also looking for testing support in the users community). So far dmq had support only for UDP, suitable to use for replication over trusted networks (local private network, or vpn). There were couple of discussions in the past about supporting other transport like tcp or tls. Earlier today I pushed several commits to dmq module trying to remove the limitation on UDP, and support the rest of the base protocols available in Kamailio (TCP, TLS and SCTP). Internally, a significant behaviour change is that the first (startup) peer notification is now done in the SIP worker one process, instead of the main process, in order to have all transports initialized. I could not spot any relevant side effect that can happen based on quick analysis, because the code with UDP only support was sending out the KDMQ requests, the corresponding replies could come back later. Anyhow, if any other dmq developer thinks of something not being quite right, reply to the sr-dev mailing list to discuss further. For users community: it would be appreciated if people using dmq can test with other transports (TLS, TCP, ...) and report any problems on sr-users mailing list or open an issue on bug tracker. Interesting test scenarios would be with replication of large amount of data or high rate of replicated requests per second. My testing got to the level of connecting to peers via TLS, trying to ensure the code is no longer limiting to UDP. I will do more, but I thought of giving a notification in early phase and get some feedback in order to avoid advancing too much in a wrong direction. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Dialog using Recod Route, with an asterisk
by Nicki Otte 15 Mar '21

15 Mar '21

Hi everyone I have a question regarding the dialog module. It uses the RecordRoute header, which seems all fine. The situation: I have a setup, where the Kamailio proxy, which is running with the dialog module, is talking to an Asterisk. Now the logic in the Asterisk, is to place a new call, and return in to the Kamailio proxy, which sends it to the PSTN. By the logic of the asterisk, it doesn’t persist with the RecordRoute headers. It starts from scratch with a new callId. Here lies the problem. Because the RecordRoute header isn’t given, then a new “dialog” is made, instead of using the one that is established. Here comes my question. Is it against the standards to make the Asterisk persist with these headers, or am I looking at the wrong solution? If not, I do see the power of the dialog module, when a call bounces between UAS’, but not when an asterisk is in the setup. Another solution could be to use a custom x- header for the dialog id, but it doesn’t look like that is a possibility. Have any of you run into this situation, and might have a solution? Thanks in advance // Nicki Bo Otte

2 2

Re: [SR-Users] Trouble with Kamailio/RTPEngine and NAT/Firewall
by PJ Slauta 15 Mar '21

15 Mar '21

Hi Jeremy, We are currently using that configuration. -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of sr-users-request(a)lists.kamailio.org Sent: Sunday, March 14, 2021 7:00 AM To: sr-users(a)lists.kamailio.org Subject: sr-users Digest, Vol 190, Issue 13 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Flists.kama… or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: iOS CallKit and tsilo (Jeremy McNamara) 2. Re: Trouble with Kamailio/RTPEngine and NAT/Firewall (Jeremy McNamara) ---------------------------------------------------------------------- Message: 1 Date: Sat, 13 Mar 2021 12:43:55 -0500 From: Jeremy McNamara <mcnamara.jeremy(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] iOS CallKit and tsilo Message-ID: <CAO0X5JSeyYHB8MhWrxd9khdQNbSigfGVOVZBw43=z0rVwhi-GA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Igor - Are you following the example from Frederico? We have implemented his idea for our iOS and Android apps... Make sure to send over the SIP CallID in the push notification. That way if a CANCEL does get sent (on JOIN/RESUME) the app can react to it correctly. https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.kamail…

1 0

iOS CallKit and tsilo
by Igor Olhovskiy 14 Mar '21

14 Mar '21

Hello! Is there any way to "store" already finished transactions in tsilo? Idea is to deliver, for example, canceled calls to the phone, when call already was answered on other device, but push notification arrive later? Major problem here, that there how it's working on iOS. On iOS phone first show you calling screen, than - app is waking and after app will register and receive invite with tsilo, it updates calling screen with CallerID and other info. But if call was canceled before, calling screen is shown, but app not receiving INVITE, so, call screen is just there for some timeout (for Linphone, for ex, it's 20 sec). Right now I've manage to do it via external SIPP call, that emulates "fake missed call", but maybe there is other way to "store" already dead transactions for some time? PS: Unfortunately, can't solve this on mobile app level. -- Regards, Igor

4 10

Trouble with Kamailio/RTPEngine and NAT/Firewall
by PJ Slauta 13 Mar '21

13 Mar '21

Hi, I have a Kamailio/RTPEngine setup with Asterisk on a couple of EC2 instances. There is a public IP and it's setup to accept any ports from our IP addresses. If I DMZ my machine behind my router, I get audio, otherwise I don't get any audio, but the SIP portion of the call connects properly. It isn't specific to my network either, it's the same for my partner on his network. I'm assuming that NAT is getting in the way, although I'm not 100% if it's that or if it's a problem opening ports. I have tried with SIP/RTP Passthrough enabled without success. Can anyone provide some guidance here? Thanks, PJ Slauta

2 1

Open separate TCP streams to the same destination
by Sebastian Damm 12 Mar '21

12 Mar '21

Hi, I have a peer that requests separate TCP connections for each registration to their server. Those registrations run through the same Kamailio instance on my side. Now what I tried is to use tcpops and the tcp_set_otcpid function, supplying a previously nonexistant TCP connection ID before relaying the register request. The command itself returns true, however, it does not do anything. The content of $conid of the reply is something totally different. So I guess, this function only works for already existing connections. Is there another way to explicitly tell Kamailio to open up another connection to a peer even though a connection to that peer is already open? Thanks in advance. Regards, Sebastian

3 3

Comparing if header has 0 value in body
by Jurijs Ivolga 11 Mar '21

11 Mar '21

Hi, I'm running Kamailio 5.3.5 I have following header "X-myheader: 0" Nevertheless it seems this is not working for me: if ($hdr(X-myheader)==0) { ..... }; What I'm doing wrong? Jurijs

3 4

Best way to ignore HTTP requests
by Ben Kaufman 11 Mar '21

11 Mar '21

I've set up a server listening on TCP recently, and notice that I'm receiving intermittent, random HTTP requests from the internet. While it would probably be a good idea to enforce a firewall rule to only allow known hosts to communicate, what would be the best way within Kamailio to ignore http requests? Would just checking $rP work? Regards, Ben Kaufman ben.kaufman(a)altigen.com<mailto:ben.kaufman@altigen.com> Director of Cloud Operations AltiGen Communications, Inc.

3 4

Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 10 Mar '21

10 Mar '21

I have one Kamailio that makes an outbound register on two Asterisk. I have a dispatcher with that same two Asterisk destinations, on INVITE message I call replace_from("", ) in order to call a internal extension and on failure route I receive 401 and call the process of authorization is OK. The problem is that if the first Asterisk fail and I get 500 on failure route I can not call replace_from() because is not valid on the block. Is there any way to make this type of authorization process after receiving a 500 on failure route of dispatcher ? Below my cfg file. Thanks in advance. Antonio Couto #!KAMAILIO#!ifdef WITH_DEBUG#!define DBGLEVEL 3#!else#!define DBGLEVEL 2#!endif # - flags# FLT_ - per transaction (message) flags# FLB_ - per branch flags#!define FLT_ACC 1#!define FLT_ACCMISSED 2#!define FLT_ACCFAILED 3#!define FLT_NATS 5 #!define FLB_NATB 6#!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0log_prefix="{$mt $hdr(CSeq) $ci} " children=8 auto_aliases=noenable_tls=noenable_sctp=no tcp_accept_no_cl=yes # SIP port listen=udp:192.168.0.201:5060# xhttp port listen=tcp:127.0.0.1:8181 loadmodule "xhttp.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "htable.so" loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 1) # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params -----# default retransmission timeout: 2sec modparam("tm", "fr_timer", 1500)# default invite retransmission timeout after 1xx: 40sec modparam("tm", "fr_inv_timer", 40000) # ----- rr params -----# set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) # account triggers (flags) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("db_text", "db_mode", 1) modparam("db_text", "emptystring", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "track_cseq_updates", 1) modparam("uac", "restore_dlg", 1) modparam("uac", "reg_db_url", "text:///etc/kamailio/dbtext") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_contact_addr", "192.168.0.201:5060") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations if (is_method("INVITE")) { route(DISPATCH); } else { sl_send_reply("501","Not Implemented"); exit; } } # Dispatch requests route[DISPATCH] { if ($si == "192.168.0.44" and $sp == 5060) { xlog("L_ALERT","from switch $si:$sp)\n"); $var(groupid) = "2"; } else{ xlog("L_ALERT","from others $si:$sp)\n"); $var(groupid) = "1"; } if(!ds_select_dst($var(groupid), "11")) { send_reply("503", "Service Unavailable"); exit; } if ($du =~ "192.168.0.210") { xlog("L_ALERT", "SCRIPT: uac_replace_from\n"); uac_replace_from("", "sip:4000@192.168.0.201:5060"); } xlog("L_ALERT", "SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { $var(ts) = t_get_status_code(); xlog("L_ALERT","ALERT: failure_route[RTF_DISPATCH]] from $si:$rp status:$var(ts)\n"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = "4000"; $avp(apass) = "1234"; $avp(arealm) = "asterisk"; if (uac_auth()) { route(RELAY); } exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if (ds_next_dst()) { xlog("L_ALERT", "retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("500","Server Internal Error"); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # jsonrpc dispatch event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } }

1 1

limits in hashtable
by Fabian Borot 10 Mar '21

10 Mar '21

Hi I have a hashtable with about 2500 elements in it. Some of the key_names are the same and in the config file and I traverse them using the index. The problem that I am having is that not all the values are loaded into memory. Lets say I have 5 entries with key_name and key_values like this: 1.2.34:5060:10.10.10 --> 0001232555690 1.2.34:5060:10.10.10 --> 0004562555691 1.2.34:5060:10.10.10 --> 0007892555692 1.2.34:5060:10.10.10 --> 0003212555693 1.2.34:5060:10.10.10 --> 0006542555694 when I do this kamcmd htable.get mytable 1.2.34:5060:10.10.10[0] kamcmd htable.get mytable 1.2.34:5060:10.10.10[1] kamcmd htable.get mytable 1.2.34:5060:10.10.10[2] kamcmd htable.get mytable 1.2.34:5060:10.10.10[3] kamcmd htable.get mytable 1.2.34:5060:10.10.10[4] I am supposed to get their values on the console but sometimes I can not get the 3rd or 4th value , instead I receive this error error: 500 - Key name doesn't exist in htable. What I have been doing so far is to delete extra/old keys from the physical hashtable and load the values again and then it works for a few days. But there is a process that adds entries to the hashtable and at some point it breaks again and I have to delete again old/unused ones. I can not use 'expire' here because the time it takes to become "old" is unknown/variable. This is how the table is defined..2^16 is more than enough to hold 2500 values. What am I missing here? modparam("htable", "htable", "a=>size=16;autoexpire=0;dbtable=htable;") I have taken pcap captures between the kamailio server and the db and I see all the values being transferred over the network when the kamailio is loading/refreshing the values from the db. Is there another limit somewhere? Also when I try to dump the values I get 'reply too big'. modparam("ctl", "binrpc_max_body_size", 10240) modparam("ctl", "binrpc_struct_max_body_size",10240) ./kamcmd htable.dump a ERROR: reply too big any assistance is highly appreciated fborot

3 2

Can't show single location via kamctl
by Jon Bonilla (Manwe) 09 Mar '21

09 Mar '21

Hi all Using version 5.4.x here. I save the registrations with use_domain 0 but I do use the domain module to do some checks. If I execute "kamctl ul" show I can see the registrations but if I execute "kamctl ul show username" I get "ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN" I understand I should use domain in case I was using it in the usrloc module but I'm not. How can I tell kamctl not to ask for domain? cheers, Jonm -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 4

Route - loose_route
by Ilie Soltanici 09 Mar '21

09 Mar '21

Hello, I have a strange issue here with the Route header not been respected by the Kamailio server. I'm getting the INVITE from the SBC-1, adding my own RR header (SBC-2), and send it to the SBC-3 This is the Invite I'm sending outside to the SBC-3 Proxy. ``` INVITE sip:+123456789@192.168.1.1:5060 SIP/2.0 Record-Route: <sip:SBC-2:5067;transport=tls;r2=on;ftag=as2b53df6f;lr> Record-Route: <sip:SBC-1;ftag=as2b53df6f;lr> ``` I'm getting back 200OK with mirrored RR, the call is established successfully. But, the problem is with the BYE message coming from the SBC-3: ``` BYE sip:+123456789@192.168.1.2:5060 SIP/2.0 ROUTE: <sip:SBC-2:5067;transport=tls;r2=on;ftag=as2b53df6f;lr>,<sip:SBC-1;ftag=as2b53df6f;lr> ``` There are 2 routes within a single header. Afaik, loose_function should remove his own Route header (SBC-2) and forward the message to the next Route (SBC-1), or to the URI if there are not other Routes. But, in my case, for some reason Kamailio on SBC-2 is sending the request directly to the URI, bypassing the SBC-1. In the logs I'm getting the following errors: *rr_do_force_send_socket(): no socket found to match second RR (sip:SBC-1)* *rr_do_force_send_socket(): second RR uri is not myself (sip:SBC-1)* To enforce the socket I was trying to use 'sn=interface', then I'm not getting anymore this error - but still, Kamailio sends the message directly to the URI, instead of SBC-1. Is there anything wrong with my config, or I am missing something with the Route headers? Kamailio Version: version: kamailio 5.4.4 (x86_64/linux) bdf4e0 Thank you.

2 1

Lookup issue with websockets contacts
by moisesynfante 09 Mar '21

09 Mar '21

I have configured Kamailio for websockets support and I'm having issues whenever I try to lookup a WS registered uac on the location table, it never finds the contact even though the contact is present when I dump the list via kamcmd. This issue only happens with WS, I tested using another user agent with a different protocol and it works fine, not sure if this is related to the fact that WS RURI doesn't really contain the AoR username. The AoR is something like 104-p4f578, but the contact's address stored is something like: sip:f0ck9vph@sjm8i3la4i7n.invalid;transport=ws; -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 1

TLS certificate unknown
by Daniel Hermann N'don 09 Mar '21

09 Mar '21

Hi guys, I am a beginner in kamailio technology. I want to use kamailio in secure mode with TLS. But when I try to connect via TLS, I get this error in terminal: Mar 08 12:29:36 KAMAILIO5 /usr/sbin/kamailio[1627]: ERROR: <core> [core/tcp_read.c:1512]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6d71a82800 r: 0x7f6d71a828e8 (-1) Mar 08 12:29:44 KAMAILIO5 /usr/sbin/kamailio[1629]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown tls.cfg : [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /etc/certs/kamailio/kamailio-key.pem certificate = /etc/certs/kamailio/kamailio-cert.pem ca_list = /etc/certs/demoCA/kamailio-cert.pem I followed this tutorial to generate the ssl keys: http://www.kamailio.net/dokuwiki/doku.php/tls:create-certificates Can someone please help me?

2 1

client TLS session reuse problem
by Rupesh Kumar 09 Mar '21

09 Mar '21

Hi, I am running kamailio 5.2.6 on centos 8 and openssl 1.1.1c and connecting and using it as a proxy sip endpoints to a legacy PBX over TLS. The connection to the backend PBX is over TLS 1.2 . Whenever kamailio initiates a TLS connection to the PBX, it uses session ID and a random session id . The server side has a bug and cannot handle the TLS session resuse apparently because of some bug/issue in caching the TLS sessions. The renegotiation and session_cache is by default turned off and I also explicitly set to 0 via modparam but kamailio would always send the session ID in the initial client hello and this is causing us trouble. Any help would be greatly appreciated. https://kamailio.org/docs/modules/5.2.x/modules/tls.html#tls.p.renegotiation Regards, RK

2 2

Core dumps/crash handlers
by Alex Balashov 09 Mar '21

09 Mar '21

For memory bugs like this ... [mem/f_malloc.c:599]: fm_free(): BUG: fm_free: bad pointer 0x1 (out of memory block!), called from tm: h_table.c: free_cell_helper(185) - aborting ... I'd love to see a core dump, but unfortunately the SIGCHLD from the SIGABRT-terminated child is intercepted and leads to a graceful shutdown: [main.c:743]: handle_sigs(): child process 16037 exited by a signal 6 Is there any way Kamailio can be set to dump core when this happens? -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

monitor uac registration status
by Davide Zanichelli 05 Mar '21

05 Mar '21

Hello is there a way to monitor (write to db) uacreg status ? Thanks You Davide

3 2

Kamailio and Asterisk in K8s
by H Yavari 05 Mar '21

05 Mar '21

Hi to all, I am going to run multiple Asterisk instances (Pods) in my Kubernetes cluster. I have a stateless Kamailio sip proxy also. We know that K8s can't handle port range for its services, So we have to use host network for Asterisk deployment. But in this mode, Asterisk pods can't connect to other pods through local IPs and you have to add a second network interface to your pods. One solution is using something like Multus that provides multi-home pods. But it has some issue in host-mode! Also, Using media proxy like RTPEngine in front of Asterisk instances doesn't make a change. You have to run RTPEngine in the host mode or just run it on a VM with multiple NIC.What is the practical solution for this scenario? Thank you. BRHossein

3 4

Re: [SR-Users] with 302 response dispatcher list .
by Devang Dhandhalya 05 Mar '21

05 Mar '21

On Thu, Mar 4, 2021 at 6:22 PM Devang Dhandhalya < devang.dhandhalya(a)ecosmob.com> wrote: > Thank you sir for giving this much method . > > but in my case i want to take Destination address from dispatcher.list > file only . and from there when every call coming i need to send response > 302 and in that contact header i need to add those destination by group id > . so any XAVP variable use in my case or any other way to solve this issue > ? > > Thanks in Advance > Devang > > On Thu, Mar 4, 2021 at 6:09 PM Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> then maybe you can keep the list of addresses in database and use >> sql_query() (from sqlops) to retrieve them and build the headers using >> $dbr(...). >> >> Or if you want caching, see the htable module. >> >> If you do not want to use a database system, see db_text to use with >> htable or just read file content with corex module. >> >> Further flexibility can be added by using other embedded scripting >> languages, e.g., app_lua. >> >> Cheers, >> Daniel >> On 04.03.21 12:31, Devang Dhandhalya wrote: >> >> Hello >> >> Right now i am not using dispatcher for routing purpose . but i add >> destination in dispatcher.list and i need to set all those list of >> destination addresses grouped by id in 302 response in contact header >> field . >> . >> Thanks in advanced >> Devang >> >> On Thu, Mar 4, 2021 at 4:43 PM Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> do you need dispatcher for providing something else? Or just to keep the >>> list of addresses grouped by id? There can be other variants if dispatcher >>> is not needed for something else. >>> >>> Cheers, >>> Daniel >>> On 04.03.21 09:19, Devang Dhandhalya wrote: >>> >>> my scenario is : i have dispatcher list >>> >>> # dispatcher destination sets (groups) >>> # >>> >>> # line format >>> # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) >>> attributes(str,opt) >>> >>> # proxies >>> 2 sip:172.19.56.21:5060 0 1 >>> 2 sip:172.19.56.22:5060 0 2 >>> >>> # gateways >>> 1 sip:192.168.0.11:5060 0 1 >>> 1 sip:192.168.0.22:5060 0 2 >>> 1 sip:192.168.0.33:5060 0 3 >>> >>> when call coming from first time client to kamailio . kamailio just need >>> to send 302 response to the client with all destination ip of group id 1 >>> priority wise in its contact header field . in easy way >>> >>> client ------------------> kamailio >>> 302 33(3) 22(2) 11(1) >>> <--------------- >>> >>> and when call coming second time from client to kamailio . kamailio need >>> to send group id 2 all destination ip in 302 response in its contact header >>> field . >>> >>> client ------------------> kamailio >>> 302 22(2) 21(1) >>> <--------------- >>> >>> can any one suggest me how to do this thing from kamailio side ? use of >>> append_branch is good option but how to configure this scenario . >>> >>> Thanks in advance >>> Devang >>> >>> *Disclaimer* >>> In addition to generic Disclaimer which you have agreed on our website, >>> any views or opinions presented in this email are solely those of the >>> originator and do not necessarily represent those of the Company or its >>> sister concerns. Any liability (in negligence, contract or otherwise) >>> arising from any third party taking any action, or refraining from taking >>> any action on the basis of any of the information contained in this email >>> is hereby excluded. >>> >>> *Confidentiality* >>> This communication (including any attachment/s) is intended only for the >>> use of the addressee(s) and contains information that is PRIVILEGED AND >>> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >>> of this communication is prohibited. Please inform originator if you have >>> received it in error. >>> >>> *Caution for viruses, malware etc.* >>> This communication, including any attachments, may not be free of >>> viruses, trojans, similar or new contaminants/malware, interceptions or >>> interference, and may not be compatible with your systems. You shall carry >>> out virus/malware scanning on your own before opening any attachment to >>> this e-mail. The sender of this e-mail and Company including its sister >>> concerns shall not be liable for any damage that may incur to you as a >>> result of viruses, incompleteness of this message, a delay in receipt of >>> this message or any other computer problems. >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>> >> *Disclaimer* >> In addition to generic Disclaimer which you have agreed on our website, >> any views or opinions presented in this email are solely those of the >> originator and do not necessarily represent those of the Company or its >> sister concerns. Any liability (in negligence, contract or otherwise) >> arising from any third party taking any action, or refraining from taking >> any action on the basis of any of the information contained in this email >> is hereby excluded. >> >> *Confidentiality* >> This communication (including any attachment/s) is intended only for the >> use of the addressee(s) and contains information that is PRIVILEGED AND >> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >> of this communication is prohibited. Please inform originator if you have >> received it in error. >> >> *Caution for viruses, malware etc.* >> This communication, including any attachments, may not be free of >> viruses, trojans, similar or new contaminants/malware, interceptions or >> interference, and may not be compatible with your systems. You shall carry >> out virus/malware scanning on your own before opening any attachment to >> this e-mail. The sender of this e-mail and Company including its sister >> concerns shall not be liable for any damage that may incur to you as a >> result of viruses, incompleteness of this message, a delay in receipt of >> this message or any other computer problems. >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Funding: https://www.paypal.me/dcmierla >> >> -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 2

RTPEngine control protocol via TCP
by Alex Balashov 04 Mar '21

04 Mar '21

Hi, Recent versions of RTPEngine (9.x+) implemented support for speaking the "NG" wire protocol over TCP, via the `--listen-tcp-ng` option. Does the Kamailio `rtpengine` module support this, and/or are there any plans to? We've run into a few scenarios where very large responses from RTPEngine can't make it back to Kamailio over the UDP protocol due to fragmentation problems in unique network settings like tunnels. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 2

with 302 response dispatcher list .
by Devang Dhandhalya 04 Mar '21

04 Mar '21

my scenario is : i have dispatcher list # dispatcher destination sets (groups) # # line format # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # proxies 2 sip:172.19.56.21:5060 0 1 2 sip:172.19.56.22:5060 0 2 # gateways 1 sip:192.168.0.11:5060 0 1 1 sip:192.168.0.22:5060 0 2 1 sip:192.168.0.33:5060 0 3 when call coming from first time client to kamailio . kamailio just need to send 302 response to the client with all destination ip of group id 1 priority wise in its contact header field . in easy way client ------------------> kamailio 302 33(3) 22(2) 11(1) <--------------- and when call coming second time from client to kamailio . kamailio need to send group id 2 all destination ip in 302 response in its contact header field . client ------------------> kamailio 302 22(2) 21(1) <--------------- can any one suggest me how to do this thing from kamailio side ? use of append_branch is good option but how to configure this scenario . Thanks in advance Devang -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 3

Segmentation fault on debian 10
by Bugaian A. Vitalie 03 Mar '21

03 Mar '21

Hello, fresh install on debian 10: Mar 3 11:42:16 teams01 kernel: [ 5771.292626] kamailio[1518]: segfault at 0 ip 00007f102bb5c206 sp 00007ffec00096e8 error 4 in libc-2.28.so [7f102bae6000+148000] Mar 3 11:42:16 teams01 kernel: [ 5771.292677] Code: 0f 1f 40 00 66 0f ef c0 66 0f ef c9 66 0f ef d2 66 0f ef db 48 89 f8 48 89 f9 48 81 e1 ff 0f 00 00 48 81 f9 cf 0f 00 00 77 6a <f3> 0f 6f 20 66 0f 74 e0 66 0f d7 d4 85 d2 74 04 0f bc c2 c3 48 83 Any things to try? Thanks. Vitalie.

3 2

topos event_route question
by Juha Heinanen 03 Mar '21

03 Mar '21

I tried to skip topos stripping based on R-URI hostpart like this: event_route[topos:msg-outgoing] { xinfo("******** at topos:msg-outgoing with $rd\n"); if ($rd == "example.com") drop; } and got this to syslog: Mar 2 18:35:59 lab /usr/bin/sip-proxy[22441]: INFO: <script>: ******** at topos:msg-outgoing with kamailio.org Where does $rd == kamailio.org come from, since R-URI hostpart of the request does not have that domain? Or is it so that $rd cannot be used in that event route? -- Juha

2 5

TOPOS (5.5) answers '404 Not Here' when b side send BYE
by 张阳 03 Mar '21

03 Mar '21

Dear Daniel: When I was using the topos module, I found that if a bye message was sent from side B, kamailio would return a 404 message. When I remove the topos module, the call is normal, attached is sip pcap. 172.18.67.50:5060 and 172.18.67.40:55066 Are different ports monitored by the same kamailio Looking forward to your reply! My cfg is too long,I will key information for cfg: modparam("ndb_redis", "server", "name=srv8;addr=127.0.0.1;port=6381;db=2") modparam("topos", "storage", "redis") modparam("topos", "db_url", "redis://127.0.0.1:6381/2") modparam("topos_redis", "serverid", "srv8") modparam("topos", "rr_update", 1) /*……*/ request_route { xlog("L_DBG","request_route message: $mb\n"); if (is_method("UPDATE")) { t_on_reply("UPDATE_SELF"); #t_on_branch("UPDATE_OUT"); } # per request initial checks route(REQINIT); #!ifdef PBX_OUTMODE route(MESSAGE); #!endif # handle requests within SIP dialogs route(WITHINDLG); route(CANCEL); # route(CHECK_REG); route(REGISTER); if (is_method("OPTIONS")) { route(REPLAY_OPTIONS); exit; } t_check_trans(); # /andle methods that don't make any sense in this environment if (!is_method("INVITE")) { xlog("L_INFO","Call $ci: Anomalous method $rm received from $si:$sp, relaying just in case\n"); sl_reply_error(); exit; } if(is_method("INVITE")){ dlgs_init("$fu","$tu","FusionSBC"); dlg_manage(); } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #From now on only INVITES #check if it's registered PBX user #!ifdef ASBCBROKER route(INVITE_PBX); #!else #!ifdef ASBC route(INVITE_PBX); #!endif #!endif route(INVITE_AUTHENTICATE); route(GET_RG_INFO); route(GET_NUMBERS); route(SHOW_CALL_INFO); #Manipulate IN ANI (Number and NOA) route(ANI_MANIPULATION); #Manipulate IN DNIS route(DNIS_MANIPULATION); route(SDP); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_DEBUG","Adding Record-Route header\n"); # NVF changed this 4/22/2014 - why remove the incoming route headers? #remove_hf("Record-Route"); record_route(); route(ROUTING); route(SEND); } /*……*/ route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing xlog("L_INFO","yang yangyang testin to not here"); if (loose_route()){ if(is_method("INFO")){ route("INFO_SELF"); } route(BYE); route(REINVITE); route(ACK); route(RELAY); } else { if ( is_method("ACK")) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 $var(obj) = $sht(call_objects=>$ci); xlog("L_INFO","Call $ci: ACK received in A-Leg, $fU@$si:$sp -----> $rU@$Ri:$Rp relaying downstreamrouteWITHINDLG\n"); t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } xlog("L_INFO","yang in to not here"); sl_send_reply("404","Not here"); } exit; } } 发送自 Windows 10 版邮件<https://go.microsoft.com/fwlink/?LinkId=550986>应用

1 0

Hunt groups with alias_db_lookup
by Adam McCarthy 02 Mar '21

02 Mar '21

Hello, I'm currently using alias_db_lookup to map inbound calls from my SIP carrier to internal endpoints e.g., 1NXXXXXXXXX(a)sip.example.com -> alice(a)sip.example.com. This works well, but I would now like to simultaneously call alice@ and bob@, e.g. alias_db_lookup(...) # somehow return alice@ and bob@ append_branch("sip:bob@sip.example.com") as described here: http://kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forking This is effectively a hunt group, where two users are called and once one user picks up the call, the second is sent a CANCEL. How would I acheive this with Kamailio? Thanks, Adam

3 3

Load htable from MongoDB
by Sergiu Pojoga 02 Mar '21

02 Mar '21

Hi there, Question: is it possible to load a htable, defined by columns, from a MongoDB collection, when a certain "column" is an object with multiple elements? For example: modparam("htable", "htable", "users=>size=10;dbtable=users;cols='FirstName,LastName,Location:Prefix';") Fails when trying to load the value of Location.Prefix { "FirstName": "John", "LastName": "Doe", "Language" : "en", "Location" : { "Country" : "USA", "City" : "New York", "Prefix" : "+1" } } Thanks in advance.

2 2

Need help in transforming an SDP string through a re.subst
by Denys Pozniak 02 Mar '21

02 Mar '21

Hello! There is a task to remove a fragment (*a=rtpmap:8 PCMA/8000#015#012*) from the SDP string generated by the rtpengine. *v=0#015#012o=kw 1614043485 1614043486 IN IP4 185.19.237.77#015#012s=kw#015#012c=IN IP4 1.1.1.1#015#012t=0 0#015#012m=audio 31828 RTP/AVP 0 8 101#015#012a=silenceSupp:off - - - -#015#012a=direction:both#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=sendrecv#015#012a=rtcp:31829#015#012a=ptime:20#015#012* The most logical option seems to be using the re.subst transformation, but so far no success. $avp(write_sdp_pv) = $(avp(write_sdp_pv){re.subst,/a=rtpmap:8 PCMA\/8000%23015%23012//s}); Сan there be an alternative or am I doing something wrong with the re.subst? -- BR, Denys Pozniak

3 3

Re: [SR-Users] [sr-dev] RTPEngine - No support for kernel packet forwarding available
by Henning Westerholt 02 Mar '21

02 Mar '21

(changing list to sr-users) Hello, Do you see an iptables rule belonging to the rtpengine module? If you do a test call, are the packet counters increased on this iptables rule? In the log message I see another error related to SRTP, not anything related to the kernel forwarding mode. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-dev <sr-dev-bounces(a)lists.kamailio.org> On Behalf Of Suresh Talasaniya Sent: Monday, March 1, 2021 11:03 AM To: sr-dev(a)lists.kamailio.org Subject: [sr-dev] RTPEngine - No support for kernel packet forwarding available Hello folks, RTPEngine - No support for kernel packet forwarding available, please find the below information/logs. Mar 1 01:46:45 sp201 rtpengine[14973]: INFO: [9ca2e714-95d0-9662-82e9-e89659024d66]: [control] Received command 'offer' from 127.0.0.1:52657<http://127.0.0.1:52657> Mar 1 01:46:45 sp201 rtpengine[14973]: NOTICE: [9ca2e714-95d0-9662-82e9-e89659024d66]: [core] Creating new call Mar 1 01:46:45 sp201 rtpengine[14973]: ERR: [9ca2e714-95d0-9662-82e9-e89659024d66]: [crypto] Failed to init DTLS connection: srtp unknown protection profile Mar 1 01:46:45 sp201 rtpengine[14973]: ERR: [9ca2e714-95d0-9662-82e9-e89659024d66]: [crypto] Failed to init DTLS connection: srtp unknown protection profile Mar 1 01:46:45 sp201 rtpengine[14973]: ERR: [9ca2e714-95d0-9662-82e9-e89659024d66]: [crypto] Failed to init DTLS connection: srtp unknown protection profile Mar 1 01:46:45 sp201 rtpengine[14973]: ERR: [9ca2e714-95d0-9662-82e9-e89659024d66]: [crypto] Failed to init DTLS connection: srtp unknown protection profile Mar 1 01:46:45 sp201 rtpengine[14973]: INFO: [9ca2e714-95d0-9662-82e9-e89659024d66]: [control] Replying to 'offer' from 127.0.0.1:52657<http://127.0.0.1:52657> (elapsed time 0.001429 sec) Mar 1 01:48:47 sp201 rtpengine[14973]: INFO: [9ca2e714-95d0-9662-82e9-e89659024d66]: [control] Received command 'delete' from 127.0.0.1:52865<http://127.0.0.1:52865> Mar 1 01:48:47 sp201 rtpengine[14973]: INFO: [9ca2e714-95d0-9662-82e9-e89659024d66]: [core] Scheduling deletion of call branch 'tW4HuJe5Hmnkd9RLpKSG' (via-branch 'z9hG4bKgRZGOZSlWGzaiX8oM6vpqSqzpiG6Tqvf0') in 30 seconds Mar 1 01:48:47 sp201 rtpengine[14973]: INFO: [9ca2e714-95d0-9662-82e9-e89659024d66]: [control] Replying to 'delete' from 127.0.0.1:52865<http://127.0.0.1:52865> (elapsed time 0.000124 sec) Mar 1 01:48:47 sp201 rtpengine[14973]: INFO: [9ca2e714-95d0-9662-82e9-e89659024d66]: [control] Received command 'delete' from 127.0.0.1:52865<http://127.0.0.1:52865> Mar 1 01:48:47 sp201 rtpengine[14973]: INFO: [9ca2e714-95d0-9662-82e9-e89659024d66]: [core] Scheduling deletion of call branch 'tW4HuJe5Hmnkd9RLpKSG' (via-branch '') in 30 seconds [root@sp201 src]# lsmod | grep xt_RTPENGINE xt_RTPENGINE 45829 5 [root@sp201 src]# lsmod Module Size Used by iptable_nat 12875 0 nf_nat_ipv4 14115 1 iptable_nat nf_nat 26583 1 nf_nat_ipv4 iptable_mangle 12695 0 camellia_aesni_avx_x86_64 27590 0 camellia_x86_64 52986 1 camellia_aesni_avx_x86_64 xts 12914 1 camellia_x86_64 nfnetlink_queue 18197 0 nfnetlink_log 17892 0 nfnetlink 14519 2 nfnetlink_log,nfnetlink_queue bluetooth 548730 0 rfkill 22391 1 bluetooth ipt_REJECT 12541 2 nf_reject_ipv4 13373 1 ipt_REJECT nf_conntrack_ipv4 15053 6 nf_defrag_ipv4 12729 1 nf_conntrack_ipv4 xt_RTPENGINE 45829 5 netconsole 14587 0 xt_conntrack 12760 5 nf_conntrack 139264 4 nf_nat,nf_nat_ipv4,xt_conntrack,nf_conntrack_ipv4 iptable_filter 12810 1 edac_mce_amd 31603 0 crc32_pclmul 13133 0 ghash_clmulni_intel 13273 0 aesni_intel 189456 0 lrw 13286 3 camellia_x86_64,aesni_intel,camellia_aesni_avx_x86_64 gf128mul 15139 2 lrw,xts glue_helper 13990 3 camellia_x86_64,aesni_intel,camellia_aesni_avx_x86_64 ppdev 17671 0 ablk_helper 13597 2 aesni_intel,camellia_aesni_avx_x86_64 cryptd 21190 3 ghash_clmulni_intel,aesni_intel,ablk_helper parport_pc 28205 0 virtio_rng 13029 0 joydev 17389 0 virtio_balloon 18015 0 parport 46395 2 ppdev,parport_pc i2c_piix4 22401 0 pcspkr 12718 0 ip_tables 27126 3 iptable_filter,iptable_mangle,iptable_nat xfs 997727 2 libcrc32c 12644 3 xfs,nf_nat,nf_conntrack ata_generic 12923 0 pata_acpi 13053 0 virtio_net 28085 0 virtio_blk 18472 4 virtio_console 28076 0 net_failover 18147 1 virtio_net failover 13374 1 net_failover ata_piix 35052 0 libata 243094 3 pata_acpi,ata_generic,ata_piix crct10dif_pclmul 14307 0 crct10dif_common 12595 1 crct10dif_pclmul crc32c_intel 22094 1 virtio_pci 22985 0 floppy 69432 0 virtio_ring 22991 6 virtio_blk,virtio_net,virtio_pci,virtio_rng,virtio_balloon,virtio_console serio_raw 13434 0 virtio 14959 6 virtio_blk,virtio_net,virtio_pci,virtio_rng,virtio_balloon,virtio_console [root@sp201 src]# kamailio -V version: kamailio 5.5.0-dev4 (x86_64/linux) fd8d0f flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: fd8d0f compiled on 03:59:17 Feb 16 2021 with gcc 4.8.5 Install the packages required to compile RTPEngine yum -y install epel-release yum -y install glib glib-devel gcc zlib zlib-devel openssl openssl-devel pcre pcre-devel libcurl libcurl-devel xmlrpc-c xmlrpc-c-devel yum -y install libevent-devel glib2-devel json-c-devel json-glib json-glib-devel gperf libpcap-devel git perl-IPC-Cmd libiptcdata-devel libiptcdata-devel hiredis hiredis-devel redis iptables-devel libwebsockets-devel yum -y install spandsp spandsp-devel rpm --import http://li.nux.ro/download/nux/RPM-GPG-KEY-nux.ro yum -y install http://li.nux.ro/download/nux/dextop/el7/x86_64/nux-dextop-release-0-5.el7.… yum repolist yum -y install ffmpeg ffmpeg-devel yum -y install bcg729 bcg729-devel wget https://download1.rpmfusion.org/free/el/rpmfusion-free-release-7.noarch.rpm rpm -i rpmfusion-free-release-7.noarch.rpm Get RTPEngine source from GitHub repository: cd /usr/src git clone https://github.com/sipwise/rtpengine.git rtpengine cd rtpengine Compile and install the daemon cd /usr/src/rtpengine/daemon/ make cp -fr rtpengine /usr/sbin/rtpengine cp -fr rtpengine /usr/local/bin/rtpengine Compile and install iptables extension cd /usr/src/rtpengine/iptables-extension make all cp -fr libxt_RTPENGINE.so /usr/lib64/xtables/. Compile and install the kernel module xt_RTPENGINE cd /usr/src/rtpengine/kernel-module make cp -fr xt_RTPENGINE.ko /lib/modules/`uname -r`/extra/xt_RTPENGINE.ko depmod -a modprobe -v xt_RTPENGINE Check if the kernel module loaded properly: lsmod | grep xt_RTPENGINE Make to load the module at the boot time: echo "# load xt_RTPENGINE module" >> /etc/modules-load.d/rtpengine.conf echo "xt_RTPENGINE" >> /etc/modules-load.d/rtpengine.conf Check if RTPEngine is accessible: ls -l /proc/rtpengine/control | grep root Should see: –w–w—-. 1 root root 0 Oct 16 10:32 /proc/rtpengine/control ls -l /proc/rtpengine/list -r–r–r–. 1 root root 0 Oct 16 10:32 /proc/rtpengine/list Install configuration file Get your external IP (with ‘ip addr’ for example) In the following command change 111.111.111.111 to your external IP echo "OPTIONS=\"-i 111.111.111.111 -n 127.0.0.1:2223<http://127.0.0.1:2223> -m 10000 -M 60000 -L 4 --log-facility=local1 --table=8 --delete-delay=0 --timeout=60 --silent-timeout=600 --final-timeout=7200 –offer-timeout=60 --num-threads=12 --tos=184 –no-fallback\"" > /etc/sysconfig/rtpengine Configure RTPEngine log echo "local1.* -/var/log/rtpengine/rtpengine.log;clean_template" >> /etc/rsyslog.conf mkdir -p /var/log/rtpengine touch /var/log/rtpengine/rtpengine.log Configure logrotate echo "/var/log/rtpengine/rtpengine.log { daily rotate 4 missingok dateext copytruncate compress }" > /etc/logrotate.d/rtpengine Disable logging to /var/log/messages perl -pi.bak -e 's#(\s+)\/var\/log\/messages#;local1.none$1/var/log/messages#' /etc/rsyslog.conf systemctl restart rsyslog Configure RTPEngine service echo "[Unit] Description=Kernel based rtp proxy After=syslog.target After=network-online.target After=iptables.service Requires=network-online.target [Service] Type=forking PIDFile=/var/run/rtpengine.pid EnvironmentFile=-/etc/sysconfig/rtpengine ExecStart=/usr/local/bin/rtpengine -p /var/run/rtpengine.pid \$OPTIONS Restart=on-abort [Install] WantedBy=multi-user.target" > /etc/systemd/system/rtpengine.service Enable and start RTPEengine systemctl enable rtpengine.service mkdir -p /var/spool/rtpengine systemctl start rtpengine Check status systemctl status rtpengine.service Check if g729 installed properly rtpengine --codecs | grep "G729" vim /etc/sysconfig/rtpengine OPTIONS="-i publicIP -n 127.0.0.1:2223<http://127.0.0.1:2223> -m 10000 -M 20000 -L 4 --log-facility=local0 --table=8 --delete-delay=0 --timeout=60 --silent-timeout=600 --final-timeout=7200 –offer-timeout=60 --num-threads=12 --tos=184 –no-fallback –log-level=7 --recording-dir=/var/spool/rtpengine/ " sudo systemctl restart rtpengine no output from below command: [root@sp201 iptables-extension]# tail -f /proc/rtpengine/0/list Can someone guide me where I am doing wrong? Thank you in advance. -- Regards, Suresh Talasaniya. Contact : +91-9724264776 Skype : suresh.talsaniya

1 0

UAC changing port for same registration
by Michal Popovic 02 Mar '21

02 Mar '21

Hello everybody, let's have a situation where UAC is behind the NAT and router suddenly change the external port. Kamailio in this case creates two location entries one with old port and one with the new one. Now UAC unregister from the new one and old location remains active until it expires and our support system that is listening to locations changes keep the agent online for other services. Is there any way besides the regular OPTIONS ping from UAS to UAC how to solve this issue? Thanks. Michal

3 2

how do you ell kamailio the targets?
by Daniel Siemens 01 Mar '21

01 Mar '21

Hi So how do I tell kamailio where my asterisk server is its to register against? Same goes for the modules like rtp engine? ------------------------------------------------------------------ This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email. ------------------------------------------------------------------

1 0

event_route when all modules have been loaded?
by Juha Heinanen 01 Mar '21

01 Mar '21

Is there a way to execute event_route when all modules have been loaded and Kamalio is ready to process requests? I found this in core cookbook, but what if there is no udp sip worker processes? event_route[core:worker-one-init] - executed by core after the first udp sip worker process executed the child_init() for all modules, before starting to process sip traffic -- Juha

3 4

unsupported sip invite request-uri format ??
by lgl_hust＠sina.com 01 Mar '21

01 Mar '21

In sip register: Request-URI: sip:ims.mnc001.mcc310.3gppnetwork.org , sip register is ok.In sip invite: Request-URI: tel:310010120073501;phone-context=ims.mnc001.mcc310.3gppnetwork.org, return 478 Unresolvable destination .I saw issue #2287 and #1173 . #2287 deals with register, and #1173 has no result.I want to know how to support such RURI in sip invite, does it need a patch in code ? If so , where and how to modified it . Should i modified kamailio.cfg such as bellows ??? I don't know how modified the kamailio.cfg . if (is_method("INVITE")) {$ru = "sip:ims.mnc008.mcc460.3gppnetwork.org"; # rewrite the "Request-URI"$du= "sip:ims.mnc008.mcc310.3gppnetwork.org"; } Thanks a lot.The following is the log:2021-02-23T03:12:08.391838+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:610]: parse_msg(): SIP Request:2021-02-23T03:12:08.391838+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:612]: parse_msg(): method:2021-02-23T03:12:08.392017+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:614]: parse_msg(): uri: tel:888555;phone-context=ims.mnc008.mcc310.3gppnetwork.org2021-02-23T03:12:08.392190+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0>2021-02-23T03:12:08.392358+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=102021-02-23T03:12:08.392556+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:171]: get_hdr_field(): [70]; uri=[tel:888555;phone-context=ims.mnc008.mcc310.3gppnetwork.org]2021-02-23T03:12:08.392732+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:174]: get_hdr_field(): to body (70)["888555"tel:888555;phone-context=ims.mnc008.mcc310.3gppnetwork.org#15#012], to tag (0)[]2021-02-23T03:12:08.392904+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:152]: get_hdr_field(): cseq : <1>2021-02-23T03:12:08.393072+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, = ; state=62021-02-23T03:12:08.393243+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, = <n/a>; state=172021-02-23T03:12:08.393419+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=52021-02-23T03:12:08.393615+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=22021-02-23T03:12:08.393814+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:500]: parse_headers(): this is the first via2021-02-23T03:12:08.393975+08:00 10-164 kamailio[238089]: DEBUG: [core/receive.c:319]: receive_msg(): --- received sip message - request - call-id: [tBfcbAf2y(a)10.45.0.101] - cseq: [1 INVITE]2021-02-23T03:12:08.394132+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=7702021-02-23T03:12:08.394289+08:00 10-164 kamailio[238089]: DEBUG: [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header2021-02-23T03:12:08.394455+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/receive.c:379]: receive_msg(): preparing to run routing scripts...2021-02-23T03:12:08.394623+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 702021-02-23T03:12:08.394782+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:247]: check_ruri_sip_version(): check_ruri_sip_version entered2021-02-23T03:12:08.394942+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:270]: check_ruri_sip_version(): check_ruri_sip_version passed2021-02-23T03:12:08.395104+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered2021-02-23T03:12:08.395274+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed2021-02-23T03:12:08.395439+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:318]: check_required_headers(): check_required_headers entered2021-02-23T03:12:08.395594+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:340]: check_required_headers(): check_required_headers passed2021-02-23T03:12:08.395748+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:348]: check_via1_header(): check via1 header2021-02-23T03:12:08.395905+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:506]: check_cseq_method(): check_cseq_method entered2021-02-23T03:12:08.396100+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:539]: check_cseq_method(): check_cseq_method passed2021-02-23T03:12:08.396261+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:548]: check_cseq_value(): check_cseq_value entered2021-02-23T03:12:08.396416+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:576]: check_cseq_value(): check_cseq_value passed2021-02-23T03:12:08.396570+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:585]: check_cl(): check_cl entered2021-02-23T03:12:08.396725+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:610]: check_cl(): check_cl passed2021-02-23T03:12:08.396882+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:622]: check_expires_value(): check_expires_value entered2021-02-23T03:12:08.397037+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:659]: check_expires_value(): no expires header found2021-02-23T03:12:08.397198+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:759]: check_parse_uris(): check_parse_uris entered2021-02-23T03:12:08.397352+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:763]: check_parse_uris(): parsing ruri2021-02-23T03:12:08.397521+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:777]: check_parse_uris(): looking up From header2021-02-23T03:12:08.397683+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:791]: check_parse_uris(): parsing From header2021-02-23T03:12:08.397846+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=qCfcbAf2021-02-23T03:12:08.398012+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=292021-02-23T03:12:08.398192+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:817]: check_parse_uris(): parsing From URI2021-02-23T03:12:08.398356+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:835]: check_parse_uris(): looking up To header2021-02-23T03:12:08.398510+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:857]: check_parse_uris(): check_parse_uris(): parsing To URI2021-02-23T03:12:08.398668+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity.c:902]: check_parse_uris(): check_parse_uris passed2021-02-23T03:12:08.398829+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 12021-02-23T03:12:08.398997+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:1034]: t_check_msg(): msg (0x7f648b13dd60) id=3/238089 global id=0/0 T start=0xffffffffffffffff2021-02-23T03:12:08.399156+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=36710, isACK=02021-02-23T03:12:08.399309+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bKrDfcbAf2yakh41vbaiuY]2021-02-23T03:12:08.399481+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:675]: t_lookup_request(): no transaction found2021-02-23T03:12:08.399669+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:1104]: t_check_msg(): msg (0x7f648b13dd60) id=3/238089 global id=3/238089 T end=(nil)2021-02-23T03:12:08.399832+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} siputils [checks.c:120]: has_totag(): no totag2021-02-23T03:12:08.399989+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 0==13 && [] == [192.168.12.99]2021-02-23T03:12:08.400145+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/socket_info.c:637]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 50602021-02-23T03:12:08.400303+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/resolve.c:1808]: str2ip6buf(): invalid name, no conversion to IP address possible2021-02-23T03:12:08.400465+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/name_alias.h:62]: grep_aliases(): matching (0::5060) vs. (0:ims.mnc008.mcc310.3gppnetwork.org:0)2021-02-23T03:12:08.400625+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/forward.c:423]: check_self(): host (0::5060) != me2021-02-23T03:12:08.400792+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:1328]: t_newtran(): msg (0x7f648b13dd60) id=3/238089 global id=3/238089 T start=(nil)2021-02-23T03:12:08.400952+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=36710, isACK=02021-02-23T03:12:08.401109+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bKrDfcbAf2yakh41vbaiuY]2021-02-23T03:12:08.401266+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_lookup.c:675]: t_lookup_request(): no transaction found2021-02-23T03:12:08.401420+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_hooks.c:336]: run_reqin_callbacks_internal(): trans=0x7f648403f798, callback type 1, id 0 entered2021-02-23T03:12:08.401590+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: 589961c517bb910457d06701844fbbd42021-02-23T03:12:08.401768+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_funcs.c:325]: t_relay_to(): new INVITE2021-02-23T03:12:08.401934+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_reply.c:614]: _reply_light(): reply sent out. buf=0x7f648b1439c0: SIP/2.0 100 trying -..., shmem=0x7f6484043b78: SIP/2.0 100 trying -2021-02-23T03:12:08.402096+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_reply.c:625]: _reply_light(): finished2021-02-23T03:12:08.402250+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} <script>: new branch [0] to tel:888555;phone-context=ims.mnc008.mcc310.3gppnetwork.org2021-02-23T03:12:08.402406+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/resolve.c:1740]: str2ipbuf(): invalid name, no conversion to IP address possible2021-02-23T03:12:08.402573+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/resolve.c:1808]: str2ip6buf(): invalid name, no conversion to IP address possible2021-02-23T03:12:08.402735+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/dns_cache.c:527]: _dns_hash_find(): (_sip._udp.(10), 33), h=9562021-02-23T03:12:08.402891+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/resolve.c:1808]: str2ip6buf(): invalid name, no conversion to IP address possible2021-02-23T03:12:08.403081+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/resolve.c:1740]: str2ipbuf(): invalid name, no conversion to IP address possible2021-02-23T03:12:08.403245+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/dns_cache.c:521]: _dns_hash_find(): invalid name, no cache lookup possible2021-02-23T03:12:08.403415+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} [core/resolve.c:1715]: sip_hostport2su(): could not resolve hostname: ""2021-02-23T03:12:08.403577+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [ut.h:309]: uri2dst2(): failed to resolve ""2021-02-23T03:12:08.403737+08:00 10-164 kamailio[238089]: ERROR: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches2021-02-23T03:12:08.403898+08:00 10-164 kamailio[238089]: DEBUG: {1 1 INVITE tBfcbAf2y(a)10.45.0.101} tm [t_funcs.c:335]: t_relay_to(): t_forward_nonack returned error -478 (-478)

2 1

more help needed, - Regisraion failed.
by Daniel Siemens 01 Mar '21

01 Mar '21

I have a asterisk server on an internal network 10.123.245.18. I have a kamailio server at 10.123.244.193 On this server is nginx coturn & rtpengine. The kamailio also has an elastic IP from AWS When I connect from a raspberry phone app servered by nginx The phone reports Registration failed The log entries are START:REGISTER from sip:32897@dnsname (IP:127.0.0.1:38014) What do I need to do to get the registration to happen? ------------------------------------------------------------------ This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email. ------------------------------------------------------------------

2 1

kamailio debug build
by Sergey Safarov 01 Mar '21

01 Mar '21

I looked at Makefiles and can see instruction # WARNING: do not add mode=debug or mode=release anymore in the Makefile, # use make mode=debug all instead. By default Kamailio is compiled with # debugging symbols in all cases (-g) and without optimization (-O0). https://github.com/kamailio/kamailio/blob/master/src/Makefile.defs#L749-L751 I do not understand how properly build Kamailio with disabled optimization. Could you let me know how to use "mode=debug"? Sergey

2 1

Parse multiple URI
by Duarte Rocha 01 Mar '21

01 Mar '21

Greetings, I have a Kamailio proxy where i'm doing some operations over multiple History-Infos and they come in two forms : Scenario 1 -> History-Info : URI1 History Info : URI2 Scenario 2 -> History-Info : URI1 , URI2. On scenario 1 i iterate between headers using $(hdr(name)[N]). However, i can't seem to find a similar tool to apply on scenario 2. In the case Kamailio doesn't have it, what would be the best way to iterate the URIS? I was thinking of parsing them using the comma ",", is that safe? Best Regards,

2 1

kamailio+rtpengine - number of calls
by marek 27 Feb '21

27 Feb '21

hi, i'm using kamailio+rtpengine as SBC. I need info about - current number of total calls - current number of incoming calls - current number of outgoing calls what's you preferred way? (kamailio snmp, statistics module,rtpengine-ctl list numsessions,...) Marek

3 4

From EXT parameter
by Ilie Soltanici 26 Feb '21

26 Feb '21

Hello, Having the following From header: *<From=sip:+123456789;ext=10@sip.pstnhub.microsoft.com:5061;user=phone>* How can I extract the only extension from it (ext=10)? Is there any variable already for that or I have to use select function? ( {s.select,index,separator} ) Thank you

3 2

webrtc setup
by Daniel Siemens 25 Feb '21

25 Feb '21

I have an asterisk server on an internal network 10.123.245.18. I have a kamailio server at 10.123.244.193 On this server is nginx coturn & rtpengine. The kamailio also has an elastic IP from AWS When I connect from a raspberry phone app served by nginx The phone reports Registration failed The log entries are START:REGISTER from sip:32897@dnsname (IP:127.0.0.1:38014) What do I need to do to get the registration to happen? In the client side using Raspberry phone is says web socket error. Thanks. ------------------------------------------------------------------ This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email. ------------------------------------------------------------------

2 2

Re: [SR-Users] No ACK from MS Teams
by Phillman25 Kyriacou 25 Feb '21

25 Feb '21

Hi Guys Thanks for all your recommendations. I did the change and rr headers only show the fqdn of Kamailio as you can see below in the 200 OK sent back to MS Teams, however still the same issue no ACK. |||||||||||||||||||| ==================== tag: snd pid: 31232 process: 6 time: 1613047849.768857 date: Thu Feb 11 14:50:49 2021 proto: tls ipv4 srcip: 82.102.42.58 srcport: 5061 dstip: 52.114.132.46 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK Via: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK4f5c9b66 Record-Route: <sip:sbc.intelligentsupport.eu:5060 ;ftag=dcea3c9546914ff8826afca13260da1c;lr=on> Record-Route: <sip:sbc.intelligentsupport.eu:5061 ;transport=tls;ftag=dcea3c9546914ff8826afca13260da1c;lr=on> Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> From: Phillip Kyriacou<sip:+35722010058@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=dcea3c9546914ff8826afca13260da1c To: <sip:+35796552237@sbc.intelligentsupport.eu:5061 ;user=phone>;tag=as63d965d3 Call-ID: 43f0f0be2fa05b408c81736dbe392aef CSeq: 1 INVITE Server: MediaGW V1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:+35796552237@82.102.42.52:5060> Content-Type: application/sdp Require: timer Content-Length: 339 v=0 o=root 601931926 601931926 IN IP4 82.102.42.52 s=ASTERISK c=IN IP4 82.102.42.52 t=0 0 m=audio 16360 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:IyrJsjVV0w5ORJQ3OJB0r8iyRZU+1lMjg4i9vzC2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv |||||||||||||||||||| ====================

6 11

Kamailio 5.4.4 extremely slow carrierroute preloading
by sr.maillists＠gmail.com 24 Feb '21

24 Feb '21

Hi Maillist! Does someone have experience with kamailio 5.4.4 and carrierroutes loading? We have about 50'000 carrierroutes, kamailio 4.3.6 loads them in about 10 seconds. Kamailio 5.4.4 needs about 10 minutes (!) to load the routes if I set the parameter modparam("carrierroute", "fetch_rows", 50000); Withous this parameter, v5.4.4 loads only 2000. - Virtual Machine is an ubuntu 20.04 with all current updates - Kamailio is installed from source - The database is postgres on a separate host I tested this with the same database and same VM, I only recompiled kamailio and made neccessary changes in the kamailio.cfg. In 4.3.6 I disabled the modules tls and jsonrpcs and added the module mi_fifo.so instead. Any Idea where to search? Did I miss something in the update-procedure? Regards Stefan

2 5

logging rd/wr queue usage?
by Andrew Chen 24 Feb '21

24 Feb '21

Hi all, I was wondering if there's a way to log rd/wr queue usage when processing a new incoming message? I am currently running 5.2.5 but couldn't find any variables in core cookbook or psudovar page. Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 2

regarding the kamailio.cfg
by Maharana, Kanhu Charana 24 Feb '21

24 Feb '21

Hi, We are getting some issue in the in sip registration. The scenario is like. Device SIP registration successful with kamalio. Device got reboot abruptly. When the device got power and switched on, device sent a sip registration again. This time the registration is failing in the kamalio. Is there any way to handle it, if the registration request received for a device which is already registered, kamalio should release the old context and start a fresh registration. Regards kanhu

2 1

Critical : Kamilio is not starting when loading Dispatcher module .
by Amul Tyagi 23 Feb '21

23 Feb '21

Dear Kamailio Team , I have installed Kamailio on Ubuntu 18.04 with SIREMIS and its working fine . But I want to use the Dispatcher module for load balancing . When enabling the dispatcher module Kamailio is not starting up and giving error in syslog. Please check below the logs for your reference. root@CICTPH01LLB01:/usr/lib/x86_64-linux-gnu/kamailio/modules# kamailio -E 0(19009) INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: 127.0.0.1:5060 udp: 192.168.180.200:5060 tcp: 127.0.0.1:5060 tcp: 192.168.180.200:5060 Aliases: tcp: CICTPH01LLB01:5060 tcp: localhost:5060 udp: CICTPH01LLB01:5060 udp: localhost:5060 0(19009) INFO: <core> [core/tcp_main.c:4735]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(19011) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(19011) INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available 0(19011) INFO: auth [auth_mod.c:347]: mod_init(): qop set, but nonce-count (nonce_count) support disabled 0(19011) INFO: <core> [main.c:2683]: main(): processes (at least): 32 - shm size: 67108864 - pkg size: 8388608 0(19011) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(19011) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(19011) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(19011) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 21(19044) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/19044 root@CICTPH01LLB01:/usr/lib/x86_64-linux-gnu/kamailio/modules# 22(19053) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) ============================================================================ ===================================== Feb 18 18:33:19 CICTPH01LLB01 systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Feb 18 18:33:19 CICTPH01LLB01 kamailio: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sct p module Feb 18 18:33:19 CICTPH01LLB01 kamailio: INFO: <core> [core/tcp_main.c:4735]: init_tcp(): using epoll_lt as the io watch method (auto detected) Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: Listening on Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: udp: 127.0.0.1:5060 Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: udp: 192.168.180.200:5060 Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: tcp: 127.0.0.1:5060 Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: tcp: 192.168.180.200:5060 Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: Aliases: Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: tcp: CICTPH01LLB01:5060 Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: tcp: localhost:5060 Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: udp: CICTPH01LLB01:5060 Feb 18 18:33:19 CICTPH01LLB01 kamailio[18815]: udp: localhost:5060 Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: auth [auth_mod.c:347]: mod_init(): qop set, but nonce-count (nonce_count) support disabled Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: <core> [main.c:2683]: main(): processes (at least): 32 - shm size: 67108864 - pkg size: 8388608 Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: ERROR: <core> [core/tcp_main.c:2863]: tcp_init(): bind(d, 0x7f81495f1efc, 16) on 127.0.0.1:5060 : Address already in use Feb 18 18:33:19 CICTPH01LLB01 /usr/sbin/kamailio[18817]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Feb 18 18:33:19 CICTPH01LLB01 systemd[1]: kamailio.service: Control process exited, code=exited status=255 Feb 18 18:33:19 CICTPH01LLB01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Feb 18 18:33:19 CICTPH01LLB01 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Feb 18 18:33:20 CICTPH01LLB01 systemd[1]: kamailio.service: Service hold-off time over, scheduling restart. Feb 18 18:33:20 CICTPH01LLB01 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 5. Feb 18 18:33:20 CICTPH01LLB01 systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. Feb 18 18:33:20 CICTPH01LLB01 systemd[1]: kamailio.service: Start request repeated too quickly. Feb 18 18:33:20 CICTPH01LLB01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Feb 18 18:33:20 CICTPH01LLB01 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Regards Amul Tyagi | Technical Head - Major Accounts & Support Globe : <tel:+63%20917%20488%200130> +63-9174880130 Skype : amultyagi365 Email : <mailto:am@lgorithmsolutions.com> am(a)lgorithmsolutions.com

5 4

Kamailio 5.4.4 segmentation fault after kamctl rpc dialplan.dump
by Julia Bo 22 Feb '21

22 Feb '21

Hello, Kamailio version 5.4.4. Kamailio segmentation fault after kamctl rpc dialplan.dump <dpid>, when dialplan includes variables in the replacement expression (repl_exp) field. Please see attached corefile. Thank you, Julia.

2 6

Kamailio Crash in TLS acceppt
by surendra p 22 Feb '21

22 Feb '21

Hi Team, Kamailio 5.2 getting crashed in tls_accept. Can someone help over here. #0 0x00007f1b135c09b2 in EVP_DecryptUpdate () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f1b135f3034 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f1b135f36a3 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #3 0x00007f1b135f4051 in RAND_DRBG_reseed () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #4 0x00007f1b135f457d in RAND_DRBG_generate () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #5 0x00007f1b135f46f1 in RAND_DRBG_bytes () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #6 0x00007f1b1378649e in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #7 0x00007f1b13777f5f in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #8 0x00007f1b13763f34 in SSL_do_handshake () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #9 0x00007f1acf246fce in tls_accept (c=0x7f1ad35fdc60, error=0x7ffc238dd1dc) at tls_server.c:411 #10 0x00007f1acf2503a3 in tls_read_f (c=0x7f1ad35fdc60, flags=0x7ffc238dd50c) at tls_server.c:1097 #11 0x0000559ac2ec9b46 in tcp_read_headers (c=0x7f1ad35fdc60, read_flags=0x7ffc238dd50c) at core/tcp_read.c:462 #12 0x0000559ac2eced1c in tcp_read_req (con=0x7f1ad35fdc60, bytes_read=0x7ffc238dd514, read_flags=0x7ffc238dd50c) at core/tcp_read.c:1348 #13 0x0000559ac2ed5442 in handle_io (fm=0x7f1b13d2cef8, events=8193, idx=-1) at core/tcp_read.c:1715 #14 0x0000559ac2ec424a in io_wait_loop_epoll (h=0x559ac3153f80 <io_w>, t=2, repeat=0) at core/io_wait.h:1073 #15 0x0000559ac2ed6df4 in tcp_receive_loop (unix_sock=75) at core/tcp_read.c:1829 #16 0x0000559ac2db8685 in tcp_init_children () at core/tcp_main.c:4802 #17 0x0000559ac2cb6867 in main_loop () at main.c:1714 #18 0x0000559ac2cbd6af in main (argc=15, argv=0x7ffc238ddb88) at main.c:2644

2 2

Problem with kamalio.
by Silvère Dekpon 22 Feb '21

22 Feb '21

Hi, how are you. I got this mail on github kamalio page. Here is my problem detail : Everytime when my vps reboot, i got a problem with my kamalio. It seems like service can't start. So i have to restart it myself by using "kamctl restart". When i enter this command, i got an error on screen. Here is error detail : ERROR: PID file /var/run/kamailio/kamailio.pid does not exist -- Kamailio start failed How can i solve it ?

2 1

Unregister with Multiple Contacts
by Charles Phillips 21 Feb '21

21 Feb '21

Happy Friday! To clean up contacts that have sent requests to unregister (Expires: 0) I am executing the following after a 200 OK response from the upstream server: reg_fetch_contacts("location", "$fu", "un_reg"); xlog("L_INFO", "Removing unregistered contact From: $si To: $tu Ruid: $ulc(un_reg=>ruid)\n"); unregister("location", "$fu", "$ulc(un_reg=>ruid)”); This produces the desired result when there is only one contact per AoR, but when multiple contacts are present, it seems to delete the first ruid (which makes sense as I am only providing the $fu). So, the question, can I select the contact by call-id or some other unique identifier specific to the desired contact that I can grab from the transaction? Reading the reg_fetch_contact and unregister docs, they do not seem to provide for selection beyond uri. - Charles

2 3

loose_route() does not find socket by name
by Juha Heinanen 21 Feb '21

21 Feb '21

I'm returning to this thread, since after some more testing, I found that loose_route() does not (at least always) detect that a pair of route headers denotes itself. In config I have defined: listen=tls:192.x.x.x:8005 name "ext_tls" modparam("rr", "sockname_mode", 1) and Route header looks like this: ROUTE: <sip:tenantX.teams.tutpro.com:8005;transport=tls;r2=on;sn=ext_tls;lr;n1;avp=savp;pm=0>,<sip:192.x.x.x:5070;transport=tcp;r2=on;lr;n1;avp=savp;pm=0> Then this code: loose_route(); if ($du != "") { xinfo("Routing in-dialog $rm <$ru> from <$fu> to <$du> based on route uri\n"); } produces to syslog: INFO: Routing in-dialog ACK <...> from <...> to <...> based on route uri -- Juha

1 3

Connection between Kamailio and redirect server
by Igor Potjevlesh 18 Feb '21

18 Feb '21

Hello ! I will have to connect my Kamailio proxies to a routing server which will receive INVITE and replies with 302 and a list of a SIP Contact where (gateways, IP, prefixes or whatever we want) the call must be routed to. I'm wondering what would be the best approach to handle with Kamalio those 302 and then proceed with the routing? For the moment, we use to work with carrierroute. But, since the redirect server will take the routing decision, I believe that carrierroute will no be useful anymore. Thank you for your inputs/experience. Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 4

Kamailio (docker) not even starting
by Nuñez, Mario 18 Feb '21

18 Feb '21

Hello, I started from scratch by getting this Docker image: https://hub.docker.com/layers/kamailio/kamailio/5.4.4-trusty/images/sha256-… Then running: docker pull kamailio/kamailio:5.4.4-trusty Then starting an container with the command: docker run -d --name mykamailio -h mykamailio -p "5060-5061:5060-5061" -p "5060:5060/udp" -p "5064-5065:5064-5065" -p "5064-5065:5064-5065/udp" -p "7000-7001:7000-7001" -p "7000:7000/udp" -e "KAMAILIO_LOG_LEVEL=info" -e "KAMAILIO_ENABLE_ROLES=websockets,message,presence_query,presence_sync,presence_notify_sync,registrar_sync" -e "RABBITMQ_HOSTS=rabbitmq.local" --cap-add IPC_LOCK --cap-add SYS_NICE --cap-add SYS_RESOURCE --cap-add NET_RAW kamailio/kamailio:5.4.4-trusty I get into container using cli and ran: Kamtcl start Then container stopped with this log: Listening on udp: 127.0.0.1:5060 udp: 172.17.0.4:5060 tcp: 127.0.0.1:5060 tcp: 172.17.0.4:5060 Aliases: tcp: mykamailio:5060 tcp: localhost:5060 0(1) INFO: <core> [core/tcp_main.c:4984]: init_tcp(): using epoll_lt as the io watch method (auto detected) udp: mykamailio:5060 udp: localhost:5060 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2854]: main(): processes (at least): 33 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 21(28) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/28 23(30) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 0(1) ALERT: <core> [main.c:780]: handle_sigs(): child process 77 exited normally, status=255 0(1) INFO: <core> [main.c:808]: handle_sigs(): terminating due to SIGCHLD 2(9) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 1(8) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 4(11) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 5(12) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 7(14) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 10(17) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 8(15) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 9(16) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 3(10) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 11(18) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 14(21) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 17(24) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 15(22) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 12(19) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 18(25) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 6(13) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 13(20) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 16(23) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 23(30) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 22(29) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 19(26) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 21(28) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 20(27) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 32(39) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 13 32(39) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 25(32) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 27(34) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 29(36) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 30(37) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 26(33) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 28(35) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 31(38) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 24(31) INFO: <core> [main.c:863]: sig_usr(): signal 15 received 0(1) ERROR: ctl [ctl.c:390]: mod_destroy(): ERROR: ctl: could not delete unix socket /var/run/kamailio//kamailio_ctl: No such file or directory (2) 0(1) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Any idea about what could be the problem? Thank you in advance This e-mail and the documents attached are confidential and intended solely for the addressee; it may also be privileged. If you receive this e-mail in error, please notify the sender immediately and destroy it. As its integrity cannot be secured on the Internet, the Atos group liability cannot be triggered for the message content. Although the sender endeavors to maintain a computer virus-free network, the sender does not warrant that this transmission is virus-free and will not be liable for any damages resulting from any virus transmitted. Este mensaje y los ficheros adjuntos pueden contener información confidencial destinada solamente a la(s) persona(s) mencionadas anteriormente y pueden estar protegidos por secreto profesional. Si usted recibe este correo electrónico por error, gracias por informar inmediatamente al remitente y destruir el mensaje. Al no estar asegurada la integridad de este mensaje sobre la red, Atos no se hace responsable por su contenido. Su contenido no constituye ningún compromiso para el grupo Atos, salvo ratificación escrita por ambas partes. Aunque se esfuerza al máximo por mantener su red libre de virus, el emisor no puede garantizar nada al respecto y no será responsable de cualesquiera daños que puedan resultar de una transmisión de virus.

2 1

FW: re I am trying to get a webrtc setup going. Here is what I have
by Daniel Siemens 17 Feb '21

17 Feb '21

I have tried to follow the instructions from the site, Kamaillo however won't startup a number of errors below. Is the version on the web rtc site specific to a version? Site is https://github.com/havfo/WEBRTC-to-SIP [root@ec2-sipproxy01 ~]# service kamailio status Redirecting to /bin/systemctl status kamailio.service â- kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/usr/lib/systemd/system/kamailio.service; enabled; vendor preset: disabled) Active: failed (Result: start-limit) since Fri 2021-02-12 14:52:06 CST; 4 days ago Main PID: 5329 (code=exited, status=255) Feb 12 14:52:05 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service: main process exited, code=exited, status=255/n/a Feb 12 14:52:05 ec2-sipproxy01.multiservice.com systemd[1]: Unit kamailio.service entered failed state. Feb 12 14:52:05 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service holdoff time over, scheduling restart. Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: start request repeated too quickly for kamailio.service Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: Unit kamailio.service entered failed state. Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. [root@ec2-sipproxy01 ~]# service kamailio start Redirecting to /bin/systemctl start kamailio.service [root@ec2-sipproxy01 ~]# journalctl -xe -- -- The start-up result is done. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12905]: ERROR: bad config file (4 errors) Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12905]: loading modules under config path: /usr/lib64/kamailio/modules/ Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service: main process exited, code=exited, status=255/n/a Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Unit kamailio.service entered failed state. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service holdoff time over, scheduling restart. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. -- Subject: Unit kamailio.service has finished shutting down -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit kamailio.service has finished shutting down. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Started Kamailio (OpenSER) - the Open Source SIP Server. -- Subject: Unit kamailio.service has finished start-up -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit kamailio.service has finished starting up. -- -- The start-up result is done. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12908]: ERROR: bad config file (4 errors) Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12908]: loading modules under config path: /usr/lib64/kamailio/modules/ Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service: main process exited, code=exited, status=255/n/a Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Unit kamailio.service entered failed state. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service holdoff time over, scheduling restart. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. -- Subject: Unit kamailio.service has finished shutting down -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit kamailio.service has finished shutting down. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: start request repeated too quickly for kamailio.service Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. -- Subject: Unit kamailio.service has failed -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit kamailio.service has failed. -- -- The result is failed. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Unit kamailio.service entered failed state. Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. 2021-02-16T15:35:13.330759-06:00 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. 2021-02-16T15:35:13.520260-06:00 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service holdoff time over, scheduling restart. 2021-02-16T15:35:13.520939-06:00 ec2-sipproxy01.multiservice.com systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. 2021-02-16T15:35:13.527280-06:00 ec2-sipproxy01.multiservice.com systemd[1]: Started Kamailio (OpenSER) - the Open Source SIP Server. 2021-02-16T15:35:13.540778-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: tls [tls_init.c:503]: init_tls_compression(): disabling compression... 2021-02-16T15:35:13.547098-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: syntax error 2021-02-16T15:35:13.547118-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: bad command 2021-02-16T15:35:13.547515-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: syntax error 2021-02-16T15:35:13.547528-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: 2021-02-16T15:35:13.547743-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 2021-02-16T15:35:13.547904-06:00 ec2-sipproxy01.multiservice.com kamailio[12908]: ERROR: bad config file (4 errors) 2021-02-16T15:35:13.548435-06:00 ec2-sipproxy01.multiservice.com kamailio[12908]: loading modules under config path: /usr/lib64/kamailio/modules/ 2021-02-16T15:35:13.555285-06:00 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service: main process exited, code=exited, status=255/n/a 2021-02-16T15:35:13.555547-06:00 ec2-sipproxy01.multiservice.com systemd[1]: Unit kamailio.service entered failed state. 2021-02-16T15:35:13.555755-06:00 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. 2021-02-16T15:35:13.769880-06:00 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service holdoff time over, scheduling restart. 2021-02-16T15:35:13.770589-06:00 ec2-sipproxy01.multiservice.com systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. 2021-02-16T15:35:13.770827-06:00 ec2-sipproxy01.multiservice.com systemd[1]: start request repeated too quickly for kamailio.service 2021-02-16T15:35:13.771068-06:00 ec2-sipproxy01.multiservice.com systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. 2021-02-16T15:35:13.771292-06:00 ec2-sipproxy01.multiservice.com systemd[1]: Unit kamailio.service entered failed state. 2021-02-16T15:35:13.771499-06:00 ec2-sipproxy01.multiservice.com systemd[1]: kamailio.service failed. 2021-02-16T15:35:38.243853-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: tls [tls_init.c:503]: init_tls_compression(): disabling compression... 2021-02-16T15:35:38.250426-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: syntax error 2021-02-16T15:35:38.250444-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: bad command 2021-02-16T15:35:38.250942-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: syntax error 2021-02-16T15:35:38.250954-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: 2021-02-16T15:35:38.251218-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 2021-02-16T15:35:47.558478-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: tls [tls_init.c:503]: init_tls_compression(): disabling compression... 2021-02-16T15:35:47.565760-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: syntax error 2021-02-16T15:35:47.565774-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: bad command 2021-02-16T15:35:47.566605-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: syntax error 2021-02-16T15:35:47.566629-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: 2021-02-16T15:35:47.566815-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 2021-02-16T15:36:00.494995-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: tls [tls_init.c:503]: init_tls_compression(): disabling compression... 2021-02-16T15:36:00.502491-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: syntax error 2021-02-16T15:36:00.502503-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: bad command 2021-02-16T15:36:00.503042-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: syntax error 2021-02-16T15:36:00.503052-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: 2021-02-16T15:36:00.503240-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 2021-02-16T15:36:05.742256-06:00 ec2-sipproxy01.multiservice.com kamailio: INFO: tls [tls_init.c:503]: init_tls_compression(): disabling compression... 2021-02-16T15:36:05.749633-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: syntax error 2021-02-16T15:36:05.749647-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 463, column 4: bad command 2021-02-16T15:36:05.750062-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 487, column 1: syntax error 2021-02-16T15:36:05.750072-06:00 ec2-sipproxy01.multiservice.com kamailio: CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /etc/kamailio/ka Int eh config file it having issues with rr module I have unloaded It in the above but is wont run 446 if (loose_route()) { 447 #!ifdef WITH_WEBSOCKETS 448 if ($du == "") { 449 if (!handle_ruri_alias()) { 450 xlog("L_ERR", "Bad alias <$ru>\n"); 451 sl_send_reply("400", "Bad Request"); 452 exit; 453 } 454 } 455 #!endif 456 route(DLGURI); 457 if (is_method("ACK")) { 458 # ACK is forwarded statelessy 459 route(NATMANAGE); 460 } else if (is_method("NOTIFY")) { 461 # Add Record-Route for in-dialog NOTIFY as per RFC 6665. 462 # record_route(); 463 } 464 route(RELAY); 465 } else { 466 if (is_method("SUBSCRIBE") && uri == myself) { 467 # in-dialog subscribe requests 468 route(PRESENCE); 469 exit; 470 } 471 if (is_method("ACK")) { 472 if (t_check_trans()) { 473 # no loose-route, but stateful ACK; 474 # must be an ACK after a 487 475 # or e.g. 404 from upstream server 476 route(RELAY); 477 exit; 478 } else { 479 # ACK without matching transaction ... ignore and d Yes I have the RR module /usr/lib64/kamailio/modules/: total 29380 -rwxr-xr-x. 1 root root 127160 Dec 14 12:23 acc_diameter.so -rwxr-xr-x. 1 root root 264912 Dec 14 12:23 acc.so -rwxr-xr-x. 1 root root 49400 Dec 14 12:23 alias_db.so -rwxr-xr-x. 1 root root 858072 Dec 14 12:23 app_jsdt.so -rwxr-xr-x. 1 root root 82264 Dec 14 12:23 async.so -rwxr-xr-x. 1 root root 94792 Dec 14 12:23 auth_db.so -rwxr-xr-x. 1 root root 168352 Dec 14 12:23 auth_diameter.so -rwxr-xr-x. 1 root root 199544 Dec 14 12:23 auth_identity.so -rwxr-xr-x. 1 root root 186696 Dec 14 12:23 auth.so -rwxr-xr-x. 1 root root 325600 Dec 14 12:23 avpops.so -rwxr-xr-x. 1 root root 128952 Dec 14 12:23 avp.so -rwxr-xr-x. 1 root root 53752 Dec 14 12:23 benchmark.so -rwxr-xr-x. 1 root root 21512 Dec 14 12:23 blst.so -rwxr-xr-x. 1 root root 94792 Dec 14 12:23 call_control.so -rwxr-xr-x. 1 root root 90384 Dec 14 12:23 call_obj.so -rwxr-xr-x. 1 root root 44544 Dec 14 12:23 cfg_db.so -rwxr-xr-x. 1 root root 33008 Dec 14 12:23 cfg_rpc.so -rwxr-xr-x. 1 root root 185136 Dec 14 12:23 cfgt.so -rwxr-xr-x. 1 root root 76744 Dec 14 12:23 cfgutils.so -rwxr-xr-x. 1 root root 147656 Dec 14 12:23 corex.so -rwxr-xr-x. 1 root root 29072 Dec 14 12:23 counters.so -rwxr-xr-x. 1 root root 325520 Dec 14 12:23 ctl.so -rwxr-xr-x. 1 root root 108048 Dec 14 12:23 db2_ops.so -rwxr-xr-x. 1 root root 371184 Dec 14 12:23 db_berkeley.so -rwxr-xr-x. 1 root root 192264 Dec 14 12:23 db_cluster.so -rwxr-xr-x. 1 root root 99168 Dec 14 12:23 db_flatstore.so -rwxr-xr-x. 1 root root 227840 Dec 14 12:23 db_mysql.so -rwxr-xr-x. 1 root root 537152 Dec 14 12:23 db_redis.so -rwxr-xr-x. 1 root root 57384 Dec 14 12:23 db_sqlite.so -rwxr-xr-x. 1 root root 247136 Dec 14 12:23 db_text.so -rwxr-xr-x. 1 root root 135240 Dec 14 12:23 db_unixodbc.so -rwxr-xr-x. 1 root root 147216 Dec 14 12:23 debugger.so -rwxr-xr-x. 1 root root 855392 Dec 14 12:23 dialog.so -rwxr-xr-x. 1 root root 408528 Dec 14 12:23 dispatcher.so -rwxr-xr-x. 1 root root 20088 Dec 14 12:23 diversion.so -rwxr-xr-x. 1 root root 87336 Dec 14 12:23 dlgs.so -rwxr-xr-x. 1 root root 219440 Dec 14 12:23 dmq.so -rwxr-xr-x. 1 root root 147600 Dec 14 12:23 domainpolicy.so -rwxr-xr-x. 1 root root 99352 Dec 14 12:23 domain.so -rwxr-xr-x. 1 root root 240576 Dec 14 12:23 drouting.so -rwxr-xr-x. 1 root root 99728 Dec 14 12:23 enum.so -rwxr-xr-x. 1 root root 32432 Dec 14 12:23 evrexec.so -rwxr-xr-x. 1 root root 103160 Dec 14 12:23 exec.so -rwxr-xr-x. 1 root root 32744 Dec 14 12:23 geoip.so -rwxr-xr-x. 1 root root 69824 Dec 14 12:23 group.so -rwxr-xr-x. 1 root root 40424 Dec 14 12:23 gzcompress.so -rwxr-xr-x. 1 root root 374784 Dec 14 12:23 htable.so -rwxr-xr-x. 1 root root 260856 Dec 14 12:23 http_async_client.so -rwxr-xr-x. 1 root root 269776 Dec 14 12:23 http_client.so -rwxr-xr-x. 1 root root 242912 Dec 14 12:23 imc.so -rwxr-xr-x. 1 root root 213968 Dec 14 12:23 ipops.so -rwxr-xr-x. 1 root root 317296 Dec 14 12:23 janssonrpcc.so -rwxr-xr-x. 1 root root 73952 Dec 14 12:23 jansson.so -rwxr-xr-x. 1 root root 106896 Dec 14 12:23 jsonrpcc.so -rwxr-xr-x. 1 root root 260304 Dec 14 12:23 jsonrpcs.so -rwxr-xr-x. 1 root root 73808 Dec 14 12:23 json.so -rwxr-xr-x. 1 root root 78568 Dec 14 12:23 keepalive.so -rwxr-xr-x. 1 root root 44384 Dec 14 12:23 kemix.so -rwxr-xr-x. 1 root root 82216 Dec 14 12:23 kex.so -rwxr-xr-x. 1 root root 15904 Dec 14 12:23 log_custom.so -rwxr-xr-x. 1 root root 28272 Dec 14 12:23 log_systemd.so -rwxr-xr-x. 1 root root 29072 Dec 14 12:23 malloc_test.so -rwxr-xr-x. 1 root root 114856 Dec 14 12:23 mangler.so -rwxr-xr-x. 1 root root 57216 Dec 14 12:23 matrix.so -rwxr-xr-x. 1 root root 32832 Dec 14 12:23 maxfwd.so -rwxr-xr-x. 1 root root 152816 Dec 14 12:23 mediaproxy.so -rwxr-xr-x. 1 root root 252792 Dec 14 12:23 mohqueue.so -rwxr-xr-x. 1 root root 108080 Dec 14 12:23 mqueue.so -rwxr-xr-x. 1 root root 194296 Dec 14 12:23 msilo.so -rwxr-xr-x. 1 root root 179264 Dec 14 12:23 msrp.so -rwxr-xr-x. 1 root root 169232 Dec 14 12:23 mtree.so -rwxr-xr-x. 1 root root 249560 Dec 14 12:23 nathelper.so -rwxr-xr-x. 1 root root 132816 Dec 14 12:23 nat_traversal.so -rwxr-xr-x. 1 root root 218632 Dec 14 12:23 ndb_redis.so -rwxr-xr-x. 1 root root 15760 Dec 14 12:23 nosip.so -rwxr-xr-x. 1 root root 61096 Dec 14 12:23 outbound.so -rwxr-xr-x. 1 root root 49648 Dec 14 12:23 path.so -rwxr-xr-x. 1 root root 69480 Dec 14 12:23 pdb.so -rwxr-xr-x. 1 root root 107288 Dec 14 12:23 pdt.so -rwxr-xr-x. 1 root root 285944 Dec 14 12:23 permissions.so -rwxr-xr-x. 1 root root 82472 Dec 14 12:23 pike.so -rwxr-xr-x. 1 root root 117192 Dec 14 12:23 pipelimit.so -rwxr-xr-x. 1 root root 49504 Dec 14 12:23 prefix_route.so -rwxr-xr-x. 1 root root 40448 Dec 14 12:23 presence_conference.so -rwxr-xr-x. 1 root root 81552 Dec 14 12:23 presence_dialoginfo.so -rwxr-xr-x. 1 root root 19616 Dec 14 12:23 presence_mwi.so -rwxr-xr-x. 1 root root 15480 Dec 14 12:23 presence_profile.so -rwxr-xr-x. 1 root root 15480 Dec 14 12:23 presence_reginfo.so -rwxr-xr-x. 1 root root 1059480 Dec 14 12:23 presence.so -rwxr-xr-x. 1 root root 214064 Dec 14 12:23 presence_xml.so -rwxr-xr-x. 1 root root 7280 Dec 14 12:23 print_lib.so -rwxr-xr-x. 1 root root 20152 Dec 14 12:23 print.so -rwxr-xr-x. 1 root root 69208 Dec 14 12:23 pua_bla.so -rwxr-xr-x. 1 root root 94096 Dec 14 12:23 pua_dialoginfo.so -rwxr-xr-x. 1 root root 40744 Dec 14 12:23 pua_json.so -rwxr-xr-x. 1 root root 110536 Dec 14 12:23 pua_reginfo.so -rwxr-xr-x. 1 root root 48504 Dec 14 12:23 pua_rpc.so -rwxr-xr-x. 1 root root 469376 Dec 14 12:23 pua.so -rwxr-xr-x. 1 root root 57128 Dec 14 12:23 pua_usrloc.so -rwxr-xr-x. 1 root root 192416 Dec 14 12:23 pua_xmpp.so -rwxr-xr-x. 1 root root 472080 Dec 14 12:23 p_usrloc.so -rwxr-xr-x. 1 root root 207928 Dec 14 12:23 pv_headers.so -rwxr-xr-x. 1 root root 872296 Dec 14 12:23 pv.so -rwxr-xr-x. 1 root root 114608 Dec 14 12:23 qos.so -rwxr-xr-x. 1 root root 103704 Dec 14 12:23 ratelimit.so -rwxr-xr-x. 1 root root 325912 Dec 14 12:23 registrar.so -rwxr-xr-x. 1 root root 564528 Dec 14 12:23 rls.so -rwxr-xr-x. 1 root root 232184 Dec 14 12:23 rr.so -rwxr-xr-x. 1 root root 32328 Dec 14 12:23 rtimer.so -rwxr-xr-x. 1 root root 369376 Dec 14 12:23 rtpengine.so -rwxr-xr-x. 1 root root 250480 Dec 14 12:23 rtpproxy.so -rwxr-xr-x. 1 root root 168568 Dec 14 12:23 sanity.so -rwxr-xr-x. 1 root root 560632 Dec 14 12:23 sca.so -rwxr-xr-x. 1 root root 170664 Dec 14 12:23 sctp.so -rwxr-xr-x. 1 root root 204168 Dec 14 12:23 sdpops.so -rwxr-xr-x. 1 root root 502880 Dec 14 12:23 seas.so -rwxr-xr-x. 1 root root 283440 Dec 14 12:23 sipcapture.so -rwxr-xr-x. 1 root root 62312 Dec 14 12:23 sipdump.so -rwxr-xr-x. 1 root root 326352 Dec 14 12:23 siptrace.so -rwxr-xr-x. 1 root root 75352 Dec 14 12:23 sipt.so -rwxr-xr-x. 1 root root 341424 Dec 14 12:23 siputils.so -rwxr-xr-x. 1 root root 121152 Dec 14 12:23 sl.so -rwxr-xr-x. 1 root root 86016 Dec 14 12:23 smsops.so -rwxr-xr-x. 1 root root 189544 Dec 14 12:23 sms.so -rwxr-xr-x. 1 root root 36864 Dec 14 12:23 speeddial.so -rwxr-xr-x. 1 root root 157576 Dec 14 12:23 sqlops.so -rwxr-xr-x. 1 root root 126992 Dec 14 12:23 ss7ops.so -rwxr-xr-x. 1 root root 90344 Dec 14 12:23 sst.so -rwxr-xr-x. 1 root root 32504 Dec 14 12:23 statistics.so -rwxr-xr-x. 1 root root 32752 Dec 14 12:23 statsc.so -rwxr-xr-x. 1 root root 37360 Dec 14 12:23 statsd.so -rwxr-xr-x. 1 root root 57016 Dec 14 12:23 stun.so -rwxr-xr-x. 1 root root 7216 Dec 14 12:23 systemdops.so -rwxr-xr-x. 1 root root 91960 Dec 14 12:23 tcpops.so -rwxr-xr-x. 1 root root 405344 Dec 14 12:23 textops.so -rwxr-xr-x. 1 root root 105624 Dec 14 12:23 textopsx.so -rwxr-xr-x. 1 root root 36680 Dec 14 12:23 timer.so -rwxr-xr-x. 1 root root 558432 Dec 14 12:23 tls.so -rwxr-xr-x. 1 root root 25208 Dec 14 12:23 tmrec.so -rwxr-xr-x. 1 root root 1369096 Dec 14 12:23 tm.so -rwxr-xr-x. 1 root root 138960 Dec 14 12:23 tmx.so -rwxr-xr-x. 1 root root 176800 Dec 14 12:23 topoh.so -rwxr-xr-x. 1 root root 184272 Dec 14 12:23 topos_redis.so -rwxr-xr-x. 1 root root 255600 Dec 14 12:23 topos.so -rwxr-xr-x. 1 root root 95336 Dec 14 12:23 tsilo.so -rwxr-xr-x. 1 root root 61832 Dec 14 12:23 uac_redirect.so -rwxr-xr-x. 1 root root 460072 Dec 14 12:23 uac.so -rwxr-xr-x. 1 root root 61640 Dec 14 12:23 uid_auth_db.so -rwxr-xr-x. 1 root root 62128 Dec 14 12:23 uid_avp_db.so -rwxr-xr-x. 1 root root 70328 Dec 14 12:23 uid_domain.so -rwxr-xr-x. 1 root root 37064 Dec 14 12:23 uid_gflags.so -rwxr-xr-x. 1 root root 40824 Dec 14 12:23 uid_uri_db.so -rwxr-xr-x. 1 root root 57776 Dec 14 12:23 uri_db.so -rwxr-xr-x. 1 root root 99880 Dec 14 12:23 userblacklist.so -rwxr-xr-x. 1 root root 397712 Dec 14 12:23 usrloc.so -rwxr-xr-x. 1 root root 119520 Dec 14 12:23 utils.so -rwxr-xr-x. 1 root root 216192 Dec 14 12:23 websocket.so -rwxr-xr-x. 1 root root 74112 Dec 14 12:23 xcap_client.so -rwxr-xr-x. 1 root root 185760 Dec 14 12:23 xcap_server.so -rwxr-xr-x. 1 root root 340544 Dec 14 12:23 xhttp_pi.so -rwxr-xr-x. 1 root root 380144 Dec 14 12:23 xhttp_prom.so -rwxr-xr-x. 1 root root 90104 Dec 14 12:23 xhttp_rpc.so -rwxr-xr-x. 1 root root 69856 Dec 14 12:23 xhttp.so -rwxr-xr-x. 1 root root 58912 Dec 14 12:23 xlog.so -rwxr-xr-x. 1 root root 166304 Dec 14 12:23 xmpp.so -rwxr-xr-x. 1 root root 135952 Dec 14 12:23 xprint.so I'm not sure but maybe the config generator is for a different version? [root@ec2-sipproxy01 ~]# rpm -qa |grep kamailio kamailio-tls-5.4.3-0.el7.centos.x86_64 kamailio-gzcompress-5.4.3-0.el7.centos.x86_64 kamailio-utils-5.4.3-0.el7.centos.x86_64 kamailio-sctp-5.4.3-0.el7.centos.x86_64 kamailio-unixodbc-5.4.3-0.el7.centos.x86_64 kamailio-5.4.3-0.el7.centos.x86_64 kamailio-xmpp-5.4.3-0.el7.centos.x86_64 kamailio-tcpops-5.4.3-0.el7.centos.x86_64 kamailio-http_async_client-5.4.3-0.el7.centos.x86_64 kamailio-sqlite-5.4.3-0.el7.centos.x86_64 kamailio-json-5.4.3-0.el7.centos.x86_64 kamailio-websocket-5.4.3-0.el7.centos.x86_64 kamailio-mysql-5.4.3-0.el7.centos.x86_64 kamailio-presence-5.4.3-0.el7.centos.x86_64 kamailio-http_client-5.4.3-0.el7.centos.x86_64 kamailio-bdb-5.4.3-0.el7.centos.x86_64 kamailio-outbound-5.4.3-0.el7.centos.x86_64 kamailio-redis-5.4.3-0.el7.centos.x86_64 kamailio-statsd-5.4.3-0.el7.centos.x86_64 kamailio-sipcapture-daemon-config-5.4.3-0.el7.centos.x86_64 kamailio-xhttp-pi-5.4.3-0.el7.centos.x86_64 kamailio-geoip-5.4.3-0.el7.centos.x86_64 kamailio-jansson-5.4.3-0.el7.centos.x86_64 kamailio-sipdump-5.4.3-0.el7.centos.x86_64 ------------------------------------------------------------------ This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email. ------------------------------------------------------------------

2 1

Cockroachdb and kamailio 5.4
by Jonathan Hunter 17 Feb '21

17 Feb '21

Hi Guys, Hope all are well? We are looking to implement kamailio with cockroachdb due to the advantages it gives us in terms of postgres management and clustering. I can see from a previous string http://sip-router.1086192.n5.nabble.com/CockroachDB-and-Kamailio-td189233.h… that someone has attempted this and even created their own branch. Has any further work been done on this, and if so has anyone got any advice/tips relating to it? As I am nervous about potentially using a branch and not a general release of kamailio moving forwards. I appreciate you cant support all database types but I would just be interested to hear people’s thoughts on the topic. Many thanks! Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

2 2

pipelimit with a kamailio cluster
by Daniel Donoghue 17 Feb '21

17 Feb '21

Hello, I have 2 stateful kamailio servers that receive incoming requests, load balanced across the 2 servers by our carrier trunk partners (outside of our control). I am looking at putting certain invites from these trunks into pipelimit to rate limit under some conditions but I have some questions: 1. How will pipelimit establish rates when requests can come in to either of my servers? 2. If pipelimit decides it is time to limit the requests, will both servers respect that limit? 3. does pipelimit rely on a db or can it use dmq, redis or other methods for cross-server communication? 4. Is there a maximum number of pipes that can be defined? Hope somebody can help me answer these questions. With every blessing, — Daniel Donoghue

3 4

haproxy and set_advertised_address()
by Sergey Safarov 16 Feb '21

16 Feb '21

haproxy sed dst_ip address from haproxy protocol https://github.com/kamailio/kamailio/blob/master/src/core/tcp_main.c#L987-L… How I can use this value inside set_advertised_address() function? Like set_advertised_address(dst_ip)

2 2

http_client/http_connect in request_route: INFO: <core> [main.c:857]: sig_usr(): signal 13 received
by Anthony Joseph Messina 16 Feb '21

16 Feb '21

With the following config # ----- http_client params ----- modparam("http_client", "httpredirect", 0) modparam("http_client", "connection_timeout", 2) modparam("http_client", "keep_connections", 1) modparam("http_client", "query_result", 0) modparam("http_client", "httpcon", "api=>https://api.example.com"); and after calling the following from request_route, $xavu(http=>rs)=http_connect("api", "/kamailio/message", "application/json", "$var(message_api)", "$xavu(http=>rb)"); if($xavu(http=>rs)==200) { jansson_get("result.code", $xavu(http=>rb), "$xavu(http=>code)"); jansson_get("result.message", $xavu(http=>rb), "$xavu(http=>message)"); sd_journal_print("LOG_DEBUG", "[$cfg(route)] Result: $xavu(http=>code), $xavu(http=>message)"); sl_send_reply("202", "Accepted"); exit; } I see the following in the log: INFO: <core> [main.c:857]: sig_usr(): signal 13 received It doesn't seem to have any negative effect, but I am wondering if it's something I should look into. If so, any pointers on where/how to begin? -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

1 0

Advice on handling SIP trunk to a Perimeta SBC
by Björn Bylander 16 Feb '21

16 Feb '21

Hi, So I’m tasked with setting up a SIP trunk to a Perimeta SBC (https://www.metaswitch.com/products/perimeta-sbc). Apparently, > Perimeta hides the topology of networks by rewriting or removing topology-sensitive information from SIP messages. By default, your Session Controller will make the following changes to SIP messages. > > The Session Controller generates new dialog identifiers (call-IDs, From tags and To tags) for each side of the call, so that it does not expose information about your core network to your access networks. > The Session Controller strips Record-Route and Route information from the message. > The Session Controller rewrites Contact and Via headers so that the source IP address is replaced with the local address of the outbound adjacency. > The Session Controller replaces the IP addresses in c= lines in SDP with the addresses it has allocated for media forwarding. For more information on this, see Media addresses and gates. My problem is this: When the Perimeta SBC sends its first INVITE (I can't say anything about any succeeding INVITEs yet) it's added a Record-Route header with the "lr" parameter included which, as far as I can see makes Kamailio think it should use loose routing which is all well I think. But when the SBC sends an ACK for the "200 OK" from the Kamailio side it doesn't include any Route headers and I think that makes it hard for Kamailio (at least with the standard script which is basically what I'm using) to know where to relay the ACK, at least it makes the t_check_trans() call in WITHINDLG after 'is_method("ACK")' return false which makes the script ignore the ACK. Any thoughts or suggestions on how to handle this? Thanks in advance, Björn Bylander

2 1

Usage of ldap/db2_ldap
by DEV 16 Feb '21

16 Feb '21

Hi! I'm trying to use the ldap / db2_ldap module in kamailio but I'm stuck because the docs seem to either be outdated or me being on the wrong track. >From what I understand, I need a single ldap.conf but there are two example files with different syntax: 1) https://github.com/kamailio/kamailio/blob/master/src/modules/db2_ldap/ldap.… 2) https://github.com/kamailio/kamailio/blob/master/src/modules/ldap/etc/ldap.… I'm already using ldap realtime for asterisk and want to switch kamailio to use the same objects by using this schema: https://github.com/kamailio/kamailio/blob/master/src/modules/db2_ldap/ser.s… When using syntax from (1) I receive this error: ERROR: ldap [ldap_mod.c:198]: mod_init(): no section found in config_file [/etc/kamailio/ldap.cfg] I double-checked my conf and it matches (1). When using (2), I got an error, telling me I left some tables unconfigured (even when mixing both syntax). What is the correct syntax to use ldap as my auth module and replace mysql auth tables? Thank you. Kind regards Kevin

3 5

Runnning kamailio (docker) with remote connection with MYSQL (docker) fails on start/restart due to socket or database connection
by Nuñez, Mario 16 Feb '21

16 Feb '21

Hi, I am running docker with this: https://hub.docker.com/layers/kamailio/kamailio/5.4.3-trusty/images/sha256-… and https://hub.docker.com/layers/mysql/library/mysql/8.0.23/images/sha256-8708… I made some configurations in Kamailio container: /etc/default/kamailio RUN_KAMAILIO=yes USER=kamailio GROUP=kamailio CFGFILE=/etc/kamailio/kamailio.cfg DUMP_CORE=yes /etc/kamailio/kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB /etc/kamailio/kamctlrc SIP_DOMAIN=kamailio.org DBENGINE=MYSQL DBHOST=172.19.0.3 DBPORT=3306 DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="kamailiorw" DBROUSER="kamailioro" DBROPW="kamailioro" DBACCESSHOST=172.19.0.5 DBROOTUSER="root" DBROOTPW="whatever" CHARSET="latin1" USERCOL="user" Then I run creation of database: /usr/sbin/kamdbctl create All was ok so I can see in MYSQL database: +------------------+------------+-----------------------+ | user | host | plugin | +------------------+------------+-----------------------+ | root | % | mysql_native_password | | kamailio | 172.19.0.3 | mysql_native_password | | kamailioro | 172.19.0.3 | mysql_native_password | | kamailio | 172.19.0.5 | mysql_native_password | | kamailioro | 172.19.0.5 | mysql_native_password | | kamailio | localhost | mysql_native_password | | kamailioro | localhost | mysql_native_password | | mysql.infoschema | localhost | caching_sha2_password | | mysql.session | localhost | caching_sha2_password | | mysql.sys | localhost | caching_sha2_password | | root | localhost | mysql_native_password | +------------------+------------+-----------------------+ At this point, if I try to restart kamailio i get an error and container will never go up again, so I had to delete it and restart the whole installation and configuration process. Log details are: ==========================================ON RESTART KAMAILIO FROM CONTAINER CONSOLE: Listening on udp: 127.0.0.1:5060 udp: 172.17.0.4:5060 tcp: 127.0.0.1:5060 tcp: 172.17.0.4:5060 Aliases: tcp: mykamailio:5060 tcp: localhost:5060 udp: mykamailio:5060 udp: localhost:5060 0(1) INFO: <core> [core/tcp_main.c:4984]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2834]: main(): processes (at least): 33 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 21(28) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/28 23(30) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 0(1) ALERT: <core> [main.c:774]: handle_sigs(): child process 205 exited normally, status=255 0(1) INFO: <core> [main.c:802]: handle_sigs(): terminating due to SIGCHLD 2(9) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 1(8) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 3(10) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 4(11) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 6(13) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 5(12) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 7(14) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 10(17) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 12(19) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 14(21) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 29(36) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 24(31) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 28(35) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 15(22) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 19(26) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 13(20) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 17(24) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 8(15) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 23(30) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 22(29) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 21(28) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 26(33) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 20(27) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 31(38) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 32(39) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 9(16) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 30(37) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 11(18) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 16(23) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 27(34) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 18(25) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 25(32) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 0(1) ERROR: ctl [ctl.c:390]: mod_destroy(): ERROR: ctl: could not delete unix socket /var/run/kamailio//kamailio_ctl: No such file or directory (2) 0(1) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized ==========================================SO CONTAINER WAS DOWN, AND THEN IF I TRY TO START CONTAINER AGAIN: Listening on udp: 127.0.0.1:5060 udp: 172.17.0.4:5060 udp: 172.19.0.5:5060 tcp: 127.0.0.1:5060 tcp: 172.17.0.4:5060 tcp: 172.19.0.5:5060 Aliases: tcp: mykamailio:5060 tcp: localhost:5060 udp: mykamailio:5060 udp: localhost:5060 0(1) INFO: <core> [core/tcp_main.c:4984]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: auth [auth_mod.c:346]: mod_init(): qop set, but nonce-count (nonce_count) support disabled 0(1) INFO: <core> [main.c:2834]: main(): processes (at least): 41 - shm size: 67108864 - pkg size: 8388608 0(1) ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) 0(1) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 0(1) ERROR: usrloc [dlist.c:852]: register_udomain(): failed to open database connection 0(1) ERROR: registrar [registrar.c:711]: domain_fixup(): failed to register domain 0(1) ERROR: <core> [core/route.c:1167]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:698 0(1) ERROR: <core> [core/rvalue.c:3869]: fix_rval_expr(): failure in cfg at line: 698 col: 22 0(1) ERROR: <core> [core/rvalue.c:3869]: fix_rval_expr(): failure in cfg at line: 698 col: 22 0(1) ERROR: <core> [core/route.c:1167]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:701 error -1 while trying to fix configuration 0(1) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized ======================================================SO NO WAY TO GET IT UP. Any idea on what could be the problem? I tried giving permissions to some folders in Kamailio docker container: chown kamailio:kamailio /usr/sbin/kam* chmod 755 /usr/sbin/kam* chown -R kamailio:kamailio /etc/kamailio chmod -R 755 /etc/kamailio chown -R root:kamailio /var/run chmod -R 755 /var/run chown -R kamailio:kamailio /run/kamailio chown -R 755 /run/kamailio But had no effect... and I do not know if there is any misconfiguration out there. Indeed I tried to configure it to use remote Postgresql and failed, so I came back to MYSQL as it looks somekind of default config. To Kamailio. Thank you in advance. This e-mail and the documents attached are confidential and intended solely for the addressee; it may also be privileged. If you receive this e-mail in error, please notify the sender immediately and destroy it. As its integrity cannot be secured on the Internet, the Atos group liability cannot be triggered for the message content. Although the sender endeavors to maintain a computer virus-free network, the sender does not warrant that this transmission is virus-free and will not be liable for any damages resulting from any virus transmitted. Este mensaje y los ficheros adjuntos pueden contener información confidencial destinada solamente a la(s) persona(s) mencionadas anteriormente y pueden estar protegidos por secreto profesional. Si usted recibe este correo electrónico por error, gracias por informar inmediatamente al remitente y destruir el mensaje. Al no estar asegurada la integridad de este mensaje sobre la red, Atos no se hace responsable por su contenido. Su contenido no constituye ningún compromiso para el grupo Atos, salvo ratificación escrita por ambas partes. Aunque se esfuerza al máximo por mantener su red libre de virus, el emisor no puede garantizar nada al respecto y no será responsable de cualesquiera daños que puedan resultar de una transmisión de virus.

1 0

Runnning kamailio (docker) with remote connection with MYSQL (docker) fails on start/restart due to socket or database connection
by Nuñez, Mario 16 Feb '21

16 Feb '21

Hi, I am running docker with this: https://hub.docker.com/layers/kamailio/kamailio/5.4.3-trusty/images/sha256-… and https://hub.docker.com/layers/mysql/library/mysql/8.0.23/images/sha256-8708… I made some configurations in Kamailio container: /etc/default/kamailio RUN_KAMAILIO=yes USER=kamailio GROUP=kamailio CFGFILE=/etc/kamailio/kamailio.cfg DUMP_CORE=yes /etc/kamailio/kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB /etc/kamailio/kamctlrc SIP_DOMAIN=kamailio.org DBENGINE=MYSQL DBHOST=172.19.0.3 DBPORT=3306 DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="kamailiorw" DBROUSER="kamailioro" DBROPW="kamailioro" DBACCESSHOST=172.19.0.5 DBROOTUSER="root" DBROOTPW="whatever" CHARSET="latin1" USERCOL="user" Then I run creation of database: /usr/sbin/kamdbctl create All was ok so I can see in MYSQL database: +------------------+------------+-----------------------+ | user | host | plugin | +------------------+------------+-----------------------+ | root | % | mysql_native_password | | kamailio | 172.19.0.3 | mysql_native_password | | kamailioro | 172.19.0.3 | mysql_native_password | | kamailio | 172.19.0.5 | mysql_native_password | | kamailioro | 172.19.0.5 | mysql_native_password | | kamailio | localhost | mysql_native_password | | kamailioro | localhost | mysql_native_password | | mysql.infoschema | localhost | caching_sha2_password | | mysql.session | localhost | caching_sha2_password | | mysql.sys | localhost | caching_sha2_password | | root | localhost | mysql_native_password | +------------------+------------+-----------------------+ At this point, if I try to restart kamailio i get an error and container will never go up again, so I had to delete it and restart the whole installation and configuration process. Log details are: ==========================================ON RESTART KAMAILIO FROM CONTAINER CONSOLE: Listening on udp: 127.0.0.1:5060 udp: 172.17.0.4:5060 tcp: 127.0.0.1:5060 tcp: 172.17.0.4:5060 Aliases: tcp: mykamailio:5060 tcp: localhost:5060 udp: mykamailio:5060 udp: localhost:5060 0(1) INFO: <core> [core/tcp_main.c:4984]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2834]: main(): processes (at least): 33 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 21(28) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/28 23(30) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 0(1) ALERT: <core> [main.c:774]: handle_sigs(): child process 205 exited normally, status=255 0(1) INFO: <core> [main.c:802]: handle_sigs(): terminating due to SIGCHLD 2(9) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 1(8) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 3(10) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 4(11) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 6(13) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 5(12) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 7(14) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 10(17) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 12(19) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 14(21) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 29(36) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 24(31) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 28(35) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 15(22) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 19(26) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 13(20) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 17(24) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 8(15) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 23(30) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 22(29) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 21(28) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 26(33) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 20(27) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 31(38) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 32(39) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 9(16) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 30(37) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 11(18) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 16(23) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 27(34) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 18(25) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 25(32) INFO: <core> [main.c:857]: sig_usr(): signal 15 received 0(1) ERROR: ctl [ctl.c:390]: mod_destroy(): ERROR: ctl: could not delete unix socket /var/run/kamailio//kamailio_ctl: No such file or directory (2) 0(1) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized ==========================================SO CONTAINER WAS DOWN, AND THEN IF I TRY TO START CONTAINER AGAIN: Listening on udp: 127.0.0.1:5060 udp: 172.17.0.4:5060 udp: 172.19.0.5:5060 tcp: 127.0.0.1:5060 tcp: 172.17.0.4:5060 tcp: 172.19.0.5:5060 Aliases: tcp: mykamailio:5060 tcp: localhost:5060 udp: mykamailio:5060 udp: localhost:5060 0(1) INFO: <core> [core/tcp_main.c:4984]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: auth [auth_mod.c:346]: mod_init(): qop set, but nonce-count (nonce_count) support disabled 0(1) INFO: <core> [main.c:2834]: main(): processes (at least): 41 - shm size: 67108864 - pkg size: 8388608 0(1) ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) 0(1) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 0(1) ERROR: usrloc [dlist.c:852]: register_udomain(): failed to open database connection 0(1) ERROR: registrar [registrar.c:711]: domain_fixup(): failed to register domain 0(1) ERROR: <core> [core/route.c:1167]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:698 0(1) ERROR: <core> [core/rvalue.c:3869]: fix_rval_expr(): failure in cfg at line: 698 col: 22 0(1) ERROR: <core> [core/rvalue.c:3869]: fix_rval_expr(): failure in cfg at line: 698 col: 22 0(1) ERROR: <core> [core/route.c:1167]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:701 error -1 while trying to fix configuration 0(1) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized ======================================================SO NO WAY TO GET IT UP. Any idea on what could be the problem? I tried giving permissions to some folders in Kamailio docker container: chown kamailio:kamailio /usr/sbin/kam* chmod 755 /usr/sbin/kam* chown -R kamailio:kamailio /etc/kamailio chmod -R 755 /etc/kamailio chown -R root:kamailio /var/run chmod -R 755 /var/run chown -R kamailio:kamailio /run/kamailio chown -R 755 /run/kamailio But had no effect... Thank you in advance. This e-mail and the documents attached are confidential and intended solely for the addressee; it may also be privileged. If you receive this e-mail in error, please notify the sender immediately and destroy it. As its integrity cannot be secured on the Internet, the Atos group liability cannot be triggered for the message content. Although the sender endeavors to maintain a computer virus-free network, the sender does not warrant that this transmission is virus-free and will not be liable for any damages resulting from any virus transmitted. Este mensaje y los ficheros adjuntos pueden contener información confidencial destinada solamente a la(s) persona(s) mencionadas anteriormente y pueden estar protegidos por secreto profesional. Si usted recibe este correo electrónico por error, gracias por informar inmediatamente al remitente y destruir el mensaje. Al no estar asegurada la integridad de este mensaje sobre la red, Atos no se hace responsable por su contenido. Su contenido no constituye ningún compromiso para el grupo Atos, salvo ratificación escrita por ambas partes. Aunque se esfuerza al máximo por mantener su red libre de virus, el emisor no puede garantizar nada al respecto y no será responsable de cualesquiera daños que puedan resultar de una transmisión de virus.

1 0

Info: Kamailio presentations later today at ClueCon TGI
by Daniel-Constantin Mierla 16 Feb '21

16 Feb '21

Hello, short note to inform that later today will be 3 sessions related to Kamailio at ClueCon TGI (online event): * a workshop about basics of SIP routing with Kamailio * a presentation about DID routing with Kamailio * a presentation about outsourcing services from FreeSwitch to Kamailio The first two are done by me, the last one by Fred Posner. The event is free to attend, by participating to a video conference or watch the youtube stream -- more details at: * https://www.cluecon.com/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

Re: [SR-Users] Advice on handling SIP trunk to a Perimeta SBC
by Björn Bylander 16 Feb '21

16 Feb '21

Hi again, I'm using Kamailio 5.3.2 by the way. Björn Bylander writes: > Hi, > > So I’m tasked with setting up a SIP trunk to a Perimeta SBC (https://www.metaswitch.com/products/perimeta-sbc). > > Apparently, > > Perimeta hides the topology of networks by rewriting or removing topology-sensitive information from SIP messages. By default, your Session Controller will make the following changes to SIP messages. > > > > The Session Controller generates new dialog identifiers (call-IDs, From tags and To tags) for each side of the call, so that it does not expose information about your core network to your access networks. > > The Session Controller strips Record-Route and Route information from the message. > > The Session Controller rewrites Contact and Via headers so that the source IP address is replaced with the local address of the outbound adjacency. > > The Session Controller replaces the IP addresses in c= lines in SDP with the addresses it has allocated for media forwarding. For more information on this, see Media addresses and gates. > > My problem is this: When the Perimeta SBC sends its first INVITE (I can't say anything about any succeeding INVITEs yet) it's added a Record-Route header with the "lr" parameter included which, as far as I can see makes Kamailio think it should use loose routing which is all well I think. But when the SBC sends an ACK for the "200 OK" from the Kamailio side it doesn't include any Route headers and I think that makes it hard for Kamailio (at least with the standard script which is basically what I'm using) to know where to relay the ACK, at least it makes the t_check_trans() call in WITHINDLG after 'is_method("ACK")' return false which makes the script ignore the ACK. > > Any thoughts or suggestions on how to handle this? > > Thanks in advance, > Björn Bylander

1 0

Kamailio v5.4.4 Released
by Daniel-Constantin Mierla 15 Feb '21

15 Feb '21

Hello, Kamailio SIP Server v5.4.4 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.4. For more details about version 5.4.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/02/kamailio-v5-4-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Releasing Kamailio v5.4.4
by Daniel-Constantin Mierla 15 Feb '21

15 Feb '21

Hello, I am considering to release Kamailio v5.4.4 sometime next week, likely on Monday, February 15, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

3 4

IMS-PSTN
by eyas barhouk 13 Feb '21

13 Feb '21

Hello dears I'm trying to build an IMS testbed based on kamailio and integrate it with the PSTN network, in order to do that I'm looking for AGCF,MGCF open source solution. Am I able to do the functionality of those nodes based on Kamailio,Asterisk or any other open source? thanks and regards Eyas

1 0

IMS-PSTN
by eyas37＠hotmail.com 12 Feb '21

12 Feb '21

1 0

basic help, In over my head in getting a webrtc up.
by Daniel Siemens 12 Feb '21

12 Feb '21

I am trying to get a webrtc setup going. Here is what I have I have asterisk server at 10.123.244.18. The webrtc works internally from the freepbx UCP application As well as the Raspberry phone allocation. This server doesn't have any nat on it all devices are on local / reouted networks. I have a Proxy server at 10.123.245.30 address. This server is located in AWS and has an elastic IP. On this server I have ngiinx that will load the raspberry phone up. What configuration do I need in kamailio and rtpengine to get this working. If I forward port 8089 in nginx to the /ws side on my asterisk server I can get a call to bridge but with no audio and the call end at 30 seconds when remote. It works internally fine. Likely beccuae the web browser can get to https://10.123.244.18:8089/ws ports fine. Thanks. ------------------------------------------------------------------ This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email. ------------------------------------------------------------------

2 1

loose_route() does not find socket by name
by Juha Heinanen 12 Feb '21

12 Feb '21

I did more tests and found that loose_route() does not find socket by name given in ;sn param: Feb 12 10:40:59 lab /usr/bin/sip-proxy[16975]: WARNING: rr [loose.c:799]: rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;sn=ext_tls;lr;n1;savp=avp;pm=0) In config I have defined: listen=tls:192.x.x.x:8007 name "ext_tls" -- Juha

2 8

Kamailio Memory Leak
by Uttam Giri 11 Feb '21

11 Feb '21

Hi, We have observed increased memory utilization in Kamailio, It increases every day, even though traffic is not increased. We are using Kamailio+RTPEngine in our setup. Phone---->Kamailio+RTPEngine------>Asterisk /etc/kamailio# kamcmd mod.stats all shm Module: core { sip_msg_shm_clone(496): 542688 msg_lump_cloner(986): 178776 build_req_buf_from_sip_req(2218): 111968 xavp_new_value(116): 22240 counters_prefork_init(212): 39680 cfg_clone_str(130): 40 cfg_shmize(217): 712 main_loop(1313): 8 init_pt(107): 8 init_pt(106): 8 init_pt(105): 4592 register_timer(995): 192 init_tcp(4962): 8192 init_tcp(4956): 32768 init_tcp(4947): 8 init_tcp(4940): 8 init_tcp(4933): 8 init_tcp(4927): 8 init_tcp(4915): 8 init_avps(90): 8 init_avps(89): 8 timer_alloc(498): 48 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 qm_shm_lock_init(1202): 8 Total: 1253464 } Module: sl { init_sl_stats_child(125): 5952 sl_startup(90): 8 init_sl_stats(110): 8 Total: 5968 } Module: siptrace { mod_init(319): 8 Total: 8 } Module: permissions { Total: 0 } Module: xlog { Total: 0 } Module: rtpengine { rtpp_function_call(2627): 1076024 shm_str_dup(779): 553016 rtpengine_hash_table_init(111): 14336 rtpengine_hash_table_init(96): 2048 rtpengine_hash_table_init(85): 1024 rtpengine_hash_table_init(76): 2048 rtpengine_hash_table_init(67): 2048 rtpengine_hash_table_init(58): 32 add_rtpengine_socks(975): 24 add_rtpengine_socks(956): 64 get_rtpp_set(870): 8 get_rtpp_set(861): 56 mod_init(1596): 8 mod_init(1589): 24 mod_init(1576): 8 mod_init(1569): 8 mod_init(1555): 16 Total: 1650792 } Module: pv { add_shvar(177): 8 add_shvar(170): 64 lock_set_alloc(66): 80 Total: 152 } Module: phonenum { Total: 0 } Module: textops { Total: 0 } Module: rtimer { Total: 0 } Module: tm { _reply_light(496): 1640 build_cell(334): 973848 relay_reply(2058): 69376 init_tm_stats_child(56): 7936 init_tmcb_lists(72): 16 init_tmcb_lists(70): 16 init_hash_table(482): 2097152 Total: 3149984 } Module: kex { pkg_proc_stats_init(79): 1488 Total: 1488 } Module: tmx { tmx_init_pretran_table(90): 256 Total: 256 } Module: utils { conf_init(552): 24 init_shmlock(134): 8 Total: 32 } Module: xhttp_prom { prom_metric_init(134): 8 Total: 8 } Module: dispatcher { ds_ht_init(89): 12288 reindex_dests(723): 768 mod_init(484): 80 ds_avl_insert(3621): 1712 pack_dest(411): 72 ds_ht_init(79): 32 ds_init_data(262): 16 ds_init_data(254): 16 mod_init(339): 8 mod_init(337): 8 ds_ping_active_init(138): 8 Total: 15008 } /etc/kamailio# kamcmd mod.stats all pkg Module: core { init_io_wait(469): 6192 init_modules(885): 16 rpc_hash_add(151): 1024 cnt_hash_add(394): 2048 add_callback(59): 168 register_select_table(458): 48 sr_wtimer_init(351): 136 cfg_new_group(79): 624 fix_sock_str(517): 1024 fix_sock_str(500): 240 add_alias(103): 96 add_alias(99): 96 init_dst_set(89): 32296 grp_hash_add(241): 512 fix_hostname(1457): 56 fix_socket_list(1666): 24 route_new_list(199): 184 fix_param(1114): 88 fix_param(1053): 312 fix_expr(554): 192 init_io_wait(516): 3224 subst_str(514): 720 mk_rval_expr2(2681): 17360 mk_elem(90): 232 mk_new_net(52): 48 tr_new(1730): 240 mk_rval_expr_v(2553): 216 yyparse(2852): 736 mk_rval_expr1(2621): 19096 mk_rval_expr_v(2540): 109152 pv_cache_add(349): 5616 pv_parse_format(1173): 4984 fix_param(1019): 2568 mk_action(118): 44416 cfg_declare(50): 3136 set_mod_param_regex(145): 88 set_mod_param_regex(134): 104 tr_table_add(1959): 360 register_module(233): 28784 register_module(216): 2808 load_module(417): 1552 sr_push_yy_state(1627): 1008 new_sock_info(348): 16 new_sock_info(339): 32 new_sock_info(327): 32 new_sock_info(304): 48 new_sock_info(299): 1200 add_alias(103): 56 add_alias(99): 64 yyparse(677): 40 yyparse(2387): 128 yyparse(658): 144 yyparse(1878): 168 get_hdr_field(116): 232 parse_headers(325): 64 subst_parser(301): 120 subst_parser(291): 1440 subst_parser(274): 640 pp_subst_add(58): 240 sr_push_yy_state(1697): 24 addstr(1445): 69072 rpc_hash_add(105): 16280 str_hash_alloc(59): 512 pv_table_add(236): 17248 pv_init_buffer(2139): 327680 pv_init_buffer(2129): 320 init_nonsip_hooks(43): 24 init_rlist(146): 40 rval_get_str(1259): 3696 route_add(124): 968 str_hash_alloc(59): 768 sr_push_yy_state(1687): 464 pp_define(1819): 568 init_counters(125): 128 cnt_hash_add(341): 27096 str_hash_alloc(59): 1280 Total: 762656 } Module: sl { sl_register_callback(514): 64 Total: 64 } Module: siptrace { mod_init(340): 384 Total: 384 } Module: permissions { get_pathname(243): 64 Total: 64 } Module: xlog { mod_init(212): 4104 xdbg_fixup_helper(498): 616 xlog_fixup_helper(533): 3720 Total: 8440 } Module: rtpengine { child_init(1964): 240 build_rtpp_socks(1791): 8 Total: 248 } Module: pv { pv_parse_xavp_name(541): 112 pv_parse_xavp_name(515): 112 tr_parse_string(2639): 120 tr_parse_string(2628): 120 tr_parse_string(2519): 104 tr_parse_string(2509): 64 tr_parse_string(2484): 64 tr_parse_string(2473): 64 add_var(65): 176 add_var(58): 344 Total: 1280 } Module: phonenum { pv_parse_phonenum_name(131): 32 sr_phonenum_add_item(99): 8 sr_phonenum_add_item(93): 304 Total: 344 } Module: textops { fixup_method(3598): 128 hname_fixup(3528): 504 Total: 632 } Module: rtimer { stm_e_param(384): 112 stm_t_param(299): 48 Total: 160 } Module: tm { Total: 0 } Module: kex { Total: 0 } Module: tmx { Total: 0 } Module: utils { Total: 0 } Module: xhttp_prom { Total: 0 } Module: dispatcher { Total: 0 } /etc/kamailio# kamailio -v version: kamailio 5.4.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 Any idea which could cause a memory leak. Does any of the module cause an issue? Thanks in Advance. Uttam

3 2

loose_route() fails on domain name
by Juha Heinanen 11 Feb '21

11 Feb '21

Using a domain name in R-R header may not be such a good idea: Feb 11 12:12:06 lab /usr/bin/sip-proxy[735]: WARNING: rr [loose.c:799] rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;pm=0;savp=avp;lr) Or is there some means to tell K, which FQDN maps to which IP address, when there are many such FQDNs? -- Juha

4 9

ds_mark_dst( ) in DISPATCHER
by alirsm 11 Feb '21

11 Feb '21

Hi, In my failover setting using Dispatcher module, I mark the GW if fails to response. OPTIONS are being sent every 30 sec: failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { ds_mark_dst("ip"); if (ds_next_dst()) { t_set_fr(0,1000); t_on_failure("DISPATCH_FAILOVER"); route(RELAY); exit; } } } However, Kamailio still recognize it as active gw ( A.A.A.A is down, no response from OPTIONS being sent) $ sudo sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:A.A.A.A:5060 FLAGS: AP PRIORITY: 0 ATTRS: } DEST: { URI: sip:B.B.B.B:5060 FLAGS: AP PRIORITY: 0 ATTRS: } } } Any help would be appreciated. AL

1 0

R-R to MS Teams Proxy
by Juha Heinanen 11 Feb '21

11 Feb '21

When request is sent from Kamailio to MS Teams SIP Proxy, the top R-R URI needs to contain FQDN of Kamailio SIP proxy instead of its IP address. Document https://skalatan.de/de/blog/kamailio-sbc-teams suggest to replace record_route(); call with record_route_preset("SBC-DNS-DOMAIN:5061;transport=tls", "SBC-IP-ADDR:5060"); That works only in a very simple case where the request came in over UDP or TCP and SIP Proxy has only one listening address, i.e., SBC-IP-ADDR. One way to solve the problem might be a new r_r function that would take FQDN of the top R-R URI as argument or introduction of a pv from where the current record_route() function would take the FQDN if it has been set. Any comments or other solutions? -- Juha

6 19

No ACK from MS Teams
by Phillman25 Kyriacou 11 Feb '21

11 Feb '21

Dear List Hope this email finds you all well. I have followed the below tutorial on how to integrate Kamailio with MS Teams https://skalatan.de/en/blog/kamailio-sbc-teams However i have been facing an issue with MS Teams with direct routing where MS Teams does not send back an ACK after a 200 OK. The current inbound call flow is the following: *MS Teams ==> Kamailio==>Asterisk* Once Asterisk answers with 200 OK we send this 200 OK back to MS Teams however, MS Teams just doesn't answer back with ACK and call drops. The below trace shows the INVITE coming from MS Teams and the 200 OK we send back. |||||||||||||||||||| ==================== tag: rcv pid: 26135 process: 23 time: 1613025911.983278 date: Thu Feb 11 08:45:11 2021 proto: tls ipv4 srcip: 52.114.132.46 srcport: 4352 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ INVITE sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: Phillip Kyriacou<sip:+357XXXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c3da9477e05d45fca31c24a155af3318 TO: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: 9cb9a2f3144e594c87bccda76791c28e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45 RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-b-euwe.pstnhub.microsoft.com:443 ;x-i=6e5254d9-79c2-4657-b54b-68791aa8e81f;x-c=9cb9a2f3144e594c87bccda76791c28e/d/10/e21a 6ff7125e4cbd91e75ec687fd4c5d> CONTENT-LENGTH: 1097 MIN-SE: 300 SUPPORTED: timer USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.2.9.1 i.USEA.7 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SESSION-EXPIRES: 3600 v=0 o=- 170239 0 IN IP4 127.0.0.1 s=session c=IN IP4 52.113.40.94 b=CT:10000000 t=0 0 m=audio 51414 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118 c=IN IP4 52.113.40.94 a=rtcp:51415 a=ice-ufrag:BwFs a=ice-pwd:sFyvnfQ9iH101E80ZKjCxi9+ a=rtcp-mux a=candidate:1 1 UDP 2130706431 52.113.40.94 51414 typ srflx raddr 10.0.137.19 rport 51414 a=candidate:1 2 UDP 2130705918 52.113.40.94 51415 typ srflx raddr 10.0.137.19 rport 51415 a=candidate:2 1 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr 10.0.137.19 rport 49152 a=candidate:2 2 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr 10.0.137.19 rport 49152 a=label:main-audio a=mid:1 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Igc2LVM9FH5kQtkRpHw99H5SB7Rd7eKzJy4gdWJg|2^31 a=sendrecv a=rtpmap:104 SILK/16000 a=rtpmap:9 G722/8000 a=rtpmap:103 SILK/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=ptime:20 |||||||||||||||||||| ==================== tag: snd pid: 26086 process: 6 time: 1613025917.726664 date: Thu Feb 11 08:45:17 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.132.46 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK Via: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45 Record-Route: <sip:SBC-IP-ADDR:5060;ftag=c3da9477e05d45fca31c24a155af3318;lr=on> Record-Route: <sip:SBC-FQDN:5061;transport=tls;ftag=c3da9477e05d45fca31c24a155af3318;lr=on> Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> From: Phillip Kyriacou<sip:+357XXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c3da9477e05d45fca31c24a155af3318 To: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone>;tag=as659924a4 Call-ID: 9cb9a2f3144e594c87bccda76791c28e CSeq: 1 INVITE Server: MediaGW V1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:357XXXXXXXX@ASTERISK_PUBLIC_IP:5060> Content-Type: application/sdp Require: timer Content-Length: 339 v=0 o=root 411266324 411266324 IN IP4 ASTERISK_PUBLIC_IP s=ASTERISK c=IN IP4 ASTERISK_PUBLIC_IP t=0 0 m=audio 14416 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9PcaAGjbemJKTERlFTcVwmLRQoDSEQPxX0L1a6RF a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv |||||||||||||||||||| Am I doing anything wrong here that MS Teams wont respond to my 200 OK? The Kamailio SBC is showing active on MS Teams and Kamailio dispatcher shows all the MS Team hubs as AP. I also can't see any errors in the log file My TLS cfg has the following: ====================== *[server:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem <http://sbc.intelligentsupport.eu/privkey.pem>certificate = /etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem <http://sbc.intelligentsupport.eu/fullchain.pem>ca_list = /etc/ssl/certs/ca-certificates.crt* *[client:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem <http://sbc.intelligentsupport.eu/privkey.pem>certificate = /etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem <http://sbc.intelligentsupport.eu/fullchain.pem>ca_list = /etc/ssl/certs/ca-certificates.crt* Thanks very much in advance! Phillip

5 6

Time elapsed since previous message.
by David Villasmil 10 Feb '21

10 Feb '21

Hello all, Is it possible to know the elapsed time since the previously received message? On outgoing calls, I.e: when i get a 180, how long did the 100 arrived? Or the INVITE... Thanks David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 11

Kamailio Split Variable
by Mihai 10 Feb '21

10 Feb '21

Hi! Can I split the variable $fd ? the $fd has the value: asterisk-wind.voicevw.lan; and i want to take the the first part - asterisk-wind any hints ?

4 3

Health Check on Dispatcher Failover
by alirsm 10 Feb '21

10 Feb '21

Hi I have setup failover on Kamailio 5.1 using Dispatcher module as below. On a round robin algorithm, INVITEs are being sent to BOTH healthy and faulty GW. When an INVITE is being sent to the faulty GW, a re-transmission will be sent after "t_set_fr" time to the healthy GW and calls are completed successfully. I was wondering if I can setup the health check correctly so the faulty IP GW can be out of round robin till the health check pass it again, i.e. no INVITE is being sent to the faulty GW as long as the health check not pass it. Your help is greatly appreciated, AL # dispatcher.list 1 sip:X.X.X.X:5060 0 0 1 sip:Y.Y.Y.Y:5060 0 0 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("503", "No destination"); exit; } t_on_failure("DISPATCH_FAILOVER"); t_set_fr(0,1000); route(RELAY); exit; } # Failure Route failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if(!ds_next_dst()) { send_reply("503", "Out of gateways"); exit; } t_on_failure("DISPATCH_FAILOVER"); t_relay(); }

1 0

http_async_client expected result in $http_rr
by Anthony Joseph Messina 10 Feb '21

10 Feb '21

https://www.kamailio.org/docs/modules/stable/modules/http_async_client says $http_rr will be the HTTP reason phrase. I'm understanding this to mean: Code: Reason: 200 OK 404 Not Found In my current 5.4 build, $http_rr returns empty (not <null> or anything else) regardless of what the server sends in the headers. Am I misunderstanding this? I'm writing an API and was hoping to do send_reply("$http_rs", "$http_rr"); -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 2

authenticated INVITE badly managed when topos is activated
by frédéric Gaisnon 10 Feb '21

10 Feb '21

Hi, Here is some more information about my problem. I think that topos impacts challenge computing. Do you have the same behaviour I observed? Do you need more information? My tests were done with kamailio 5.4.3 on Centos7 Without topos activated (note that with topoh activated I have the same good behaviour): CPE - INVITE -> SBC CPE <- 407 ---- SBC CPE - INVITE ->SBC (with proxy-authorization header) -- INVITE --> PROXY (So in this case challenge is validated and INVITE is forwarded) With topos activated: CPE - INVITE -> SBC CPE <- 407 ---- SBC CPE - INVITE ->SBC (with proxy-authorization header) CPE <-407 -----SBC topos configuration: loadmodule "ndb_redis.so" loadmodule "topos.so" loadmodule "topos_redis.so" # ----- topos params ----- modparam("topos", "storage", "redis") modparam("topos", "dialog_expire", 15000) Code used: # IP authorization and user authentication route[AUTH] { xlog("L_DBG", "route[AUTH]\n"); #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif #!ifdef WITH_AUTH if ((is_method("REGISTER")) || ($avp(need_auth) == "1")) { ####need_auth is equal to 1 in this case # authenticate requests $var(key)=$fU + "@" + $fd; if($sht(auth_cache=>$var(key))!=$null) { if (!pv_auth_check("$fd", "$sht(auth_cache=>$var(key))", "0", "1")) { auth_challenge("$fd", “1”); #################### we always go here with INVITE with proxy-authorization header and the return code is always -5 (AUTH_NO_CREDENTIALS) exit; } } else { if (!auth_check("$fd", "subscriber", "1")) { if ($rc == -1) { append_to_reply("Retry-After: 10\r\n"); send_reply("503", "Authentication server error"); exit; } auth_challenge("$fd", "0"); exit; } $sht(auth_cache=>$var(key)) = $avp(password); } # user authenticated - remove auth header consume_credentials(); ######## without topos we go here with INVITE with proxy-authorization header } #!endif return; } Note that in this case (with topos) the return code of function pv_auth_check is always -5 (AUTH_NO_CREDENTIALS) CASE OK: Frame 3279: 545 bytes on wire (4360 bits), 545 bytes captured (4360 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.102, Dst: 192.168.1.11 Transmission Control Protocol, Src Port: 5060, Dst Port: 60796, Seq: 1, Ack: 953, Len: 477 Session Initiation Protocol (407) Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK2df8e195D1847B94;rport=60796;received=192.168.1.11 From: "6200" <sip:6200@entreprise-108.fr>;tag=B583B663-FBFBFCAA To: <sip:0900000000@entreprise-108.fr ;user=phone>;tag=83518db21d5b2e9b777975024049f5a3.8f270000 CSeq: 1 INVITE Call-ID: 9378ee27e6b7aea384a881c938de8138 [Generated Call-ID: 9378ee27e6b7aea384a881c938de8138] Proxy-Authenticate: Digest realm="entreprise-108.fr", nonce="YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Authentication Scheme: Digest Realm: "entreprise-108.fr" Nonce Value: "YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Content-Length: 0 Frame 3285: 1259 bytes on wire (10072 bits), 1259 bytes captured (10072 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.102 Transmission Control Protocol, Src Port: 60796, Dst Port: 5060, Seq: 1578, Ack: 478, Len: 1191 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0900000000@entreprise-108.fr;user=phone;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK827c83577BAADACE From: "6200" <sip:6200@entreprise-108.fr>;tag=B583B663-FBFBFCAA SIP Display info: "6200" SIP from address: sip:6200@entreprise-108.fr SIP from tag: B583B663-FBFBFCAA To: <sip:0900000000@entreprise-108.fr;user=phone> SIP to address: sip:0900000000@entreprise-108.fr;user=phone CSeq: 2 INVITE Call-ID: 9378ee27e6b7aea384a881c938de8138 [Generated Call-ID: 9378ee27e6b7aea384a881c938de8138] Contact: <sip:6200@192.168.1.33;transport=tcp> Contact URI: sip:6200@192.168.1.33;transport=tcp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_500-UA/5.7.0.14430 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Supported: replaces,100rel Allow-Events: conference,talk,hold Proxy-Authorization: Digest username="6200", realm=" entreprise-108.fr", nonce="YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO", uri=" sip:0900000000@entreprise-108.fr;user=phone;transport=tcp", response="3e0013cc3dc3855602ce1939af7e6f40", algorithm=MD5 Authentication Scheme: Digest Username: "6200" Realm: "entreprise-108.fr" Nonce Value: "YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Authentication URI: "sip:0900000000@entreprise-108.fr ;user=phone;transport=tcp" Digest Authentication Response: "3e0013cc3dc3855602ce1939af7e6f40" Algorithm: MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 270 Message Body Bad case (with topos activated): Frame 9071: 545 bytes on wire (4360 bits), 545 bytes captured (4360 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.102, Dst: 192.168.1.11 Transmission Control Protocol, Src Port: 5060, Dst Port: 43608, Seq: 1, Ack: 953, Len: 477 Session Initiation Protocol (407) Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK5c0a58f3707458FA;rport=43608;received=192.168.1.11 From: "6200" <sip:6200@entreprise-108.fr>;tag=59191351-FD3B2D60 To: <sip:0900000000@entreprise-108.fr ;user=phone>;tag=83518db21d5b2e9b777975024049f5a3.8f270000 CSeq: 1 INVITE Call-ID: 727c871081e29672abcb8bd05dde8138 [Generated Call-ID: 727c871081e29672abcb8bd05dde8138] Proxy-Authenticate: Digest realm="entreprise-108.fr", nonce="YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Authentication Scheme: Digest Realm: "entreprise-108.fr" Nonce Value: "YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Content-Length: 0 Frame 9078: 1259 bytes on wire (10072 bits), 1259 bytes captured (10072 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.102 Transmission Control Protocol, Src Port: 43608, Dst Port: 5060, Seq: 1578, Ack: 478, Len: 1191 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0900000000@entreprise-108.fr;user=phone;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bKbca400a5DCDB8264 From: "6200" <sip:6200@entreprise-108.fr>;tag=59191351-FD3B2D60 SIP Display info: "6200" SIP from address: sip:6200@entreprise-108.fr SIP from tag: 59191351-FD3B2D60 To: <sip:0900000000@entreprise-108.fr;user=phone> SIP to address: sip:0900000000@entreprise-108.fr;user=phone CSeq: 2 INVITE Call-ID: 727c871081e29672abcb8bd05dde8138 [Generated Call-ID: 727c871081e29672abcb8bd05dde8138] Contact: <sip:6200@192.168.1.33;transport=tcp> Contact URI: sip:6200@192.168.1.33;transport=tcp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_500-UA/5.7.0.14430 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Supported: replaces,100rel Allow-Events: conference,talk,hold Proxy-Authorization: Digest username="6200", realm=" entreprise-108.fr", nonce="YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/", uri=" sip:0900000000@entreprise-108.fr;user=phone;transport=tcp", response="281d775e7166a96d5efe2e100df3df9a", algorithm=MD5 Authentication Scheme: Digest Username: "6200" Realm: "entreprise-108.fr" Nonce Value: "YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Authentication URI: "sip:0900000000@entreprise-108.fr ;user=phone;transport=tcp" Digest Authentication Response: "281d775e7166a96d5efe2e100df3df9a" Algorithm: MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 270 Message Body Regards, Frederic

1 0

CANCEL transaction
by David Villasmil 10 Feb '21

10 Feb '21

Hello, When receiving a reply, there's a logic which if true, i must cancel the outgoing invite (can 480 the origination). I tried using something like t_set_fr(10,10); to have kamailio cancel it immediately, but it's not happening. Is there a way to cancel an INVITE/transaction? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 7

master daily build with disabled compiler optimisation
by Sergey Safarov 10 Feb '21

10 Feb '21

I want suggested to use "-ggdb3 -O0" gcc options when compiled daily master RPM files. Through?

1 0

Permissions module with Mongodb
by H Yavari 09 Feb '21

09 Feb '21

Hi to all, It's mentioned in the Permissions module doc, we can set DB driver:https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#i… So we can use MongoDB as a DB backend? All the modules that use DB, support Mongo? Thanks. Regards,Hossein

4 13

Kamailio as MS-Teams SBC
by Davide Zanichelli 09 Feb '21

09 Feb '21

Hello everybody! i am trying a setup of kamailio like sbc for ms teams, I currently have the following scenario: TLS Connection OK Out Call ok (except BYE 481 on reply) my problem is in incoming calls microsoft only supports srtp and my pbx (freeswitch based) only works with rtp, now i was reading that you need to use rtpengine to manage srtp to rtp, unfortunately it seems that rtpengine has become very difficult to build especially with recent os, I was wondering if it is possible to have an alternative to rtpengine for managing the srtp (rtpproxy does not seem to work) has anyone managed to get a working setup? in case it is possible to send a configuration template? Thanks You !!! Davide

8 26

run_trans_callbacks_internal crash
by Juha Heinanen 09 Feb '21

09 Feb '21

We got the crash below with 5.4.2. Before the crash there was db_mysql_async_exec_task(): failed to execute query related error messages in syslog. -- Juha Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f601d4b2f32 in run_trans_callbacks_internal (cb_lst=0x7f5ffb53e318, type=512, trans=0x7f5ffb53e2a0, params=0x7ffd1c349720) at t_hooks.c:254 254 t_hooks.c: No such file or directory. (gdb) bt full #0 0x00007f601d4b2f32 in run_trans_callbacks_internal (cb_lst=0x7f5ffb53e318, type=512, trans=0x7f5ffb53e2a0, params=0x7ffd1c349720) at t_hooks.c:254 cbp = 0x72743b6d6f632e61 backup_from = 0x560e8abbe2f0 <def_list+16> backup_to = 0x560e8abbe2f8 <def_list+24> backup_dom_from = 0x560e8abbe300 <def_list+32> backup_dom_to = 0x560e8abbe308 <def_list+40> backup_uri_from = 0x560e8abbe2e0 <def_list> backup_uri_to = 0x560e8abbe2e8 <def_list+8> backup_xavps = 0x560e8abbeae0 <_xavp_list_head> backup_xavus = 0x560e8abbeae8 <_xavu_list_head> backup_xavis = 0x560e8abbeaf0 <_xavi_list_head> __func__ = "run_trans_callbacks_internal" #1 0x00007f601d4b35bb in run_trans_callbacks_with_buf (type=512, rbuf=0x7f5ffb53e370, req=0x7f5ffafe3b70, repl=0xffffffffffffffff, flags=0) at t_hooks.c:303 params = {req = 0x7f5ffafe3b70, rpl = 0xffffffffffffffff, param = 0x7f5ffa61e640, code = 487, flags = 0, branch = 0, t_rbuf = 0x7f5ffb53e370, dst = 0x7f5ffb53e3c0, send_buf = {s = 0x7f5ffa6d4d40 "F", len = 621}} trans = 0x7f5ffb53e2a0 #2 0x00007f601d458025 in relay_reply (t=0x7f5ffb53e2a0, p_msg=0xffffffffffffffff, branch=0, msg_status=408, cancel_data=0x7ffd1c349910, do_put_on_wait=0) at t_reply.c:2124 relay = 0 save_clone = 0 buf = 0x7f601e46a4e0 "SIP/2.0 487 Request Terminated\r\nVia: SIP/2.0/TCP 46.182.160.40;branch=z9hG4bK9ef2.1bbf20df35272cd85c43814302770601.2\r\nVia: SIP/2.0/UDP 95.216.75.211;rport=5060;branch=z9hG4bK9ef2.d6265991eb8da9b735273"... res_len = 621 relayed_code = 487 relayed_msg = 0xffffffffffffffff reply_bak = 0x20000000 bm = {to_tag_val = { s = 0x7f601e46a692 "f148edd50b878efc51a72f445b390303-e1110000\r\nCall-ID: c2e2b55c-e574-1239-3cbd-6cb3112351ce\r\nCSeq: 31882977 INVITE\r\nServer: OpenSIPg SIP Proxy (5.4.2-8 (x86_64/linux))\r\nContent-Length: 0\r\n\r\n", len = 41}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7f5ffb53e370 to_tag = 0x7f601d525770 <tm_tag> reason = {s = 0x560e8aad2dbb "Request Terminated", len = 18} onsend_params = {req = 0x7ffd1c349940, rpl = 0x7f5ffaeeb7c0, param = 0x7ffd1c349880, code = 491640932, flags = 32608, branch = 0, t_rbuf = 0x59d, dst = 0x7f5ffb0ea590, send_buf = {s = 0x7ffd1c349940 "p\231\064\034\375\177", len = 491699179}} ip = {af = 1, len = 0, u = {addrl = {140050215200048, 140050209586032}, addr32 = {4216579376, 32607, 4210965360, 32607}, addr16 = {58672, 64339, 32607, 0, 15216, 64254, 32607, 0}, addr = "0\345S\373_\177\000\000p;\376\372_\177\000"}} __func__ = "relay_reply" #3 0x00007f601d49c7b3 in fake_reply (t=0x7f5ffb53e2a0, branch=0, code=408) at timer.c:295 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 517217456}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 517217456}}}} do_cancel_branch = 0 reply_status = 3845367323 #4 0x00007f601d49cc35 in final_response_handler (r_buf=0x7f5ffb53e540, t=0x7f5ffb53e2a0) at timer.c:462 silent = 0 branch_ret = -1 prev_branch = -1 now = 2146494577 #5 0x00007f601d49cd29 in retr_buf_handler (ticks=2146494577, tl=0x7f5ffb53e560, p=0xfa0) at timer.c:518 rbuf = 0x7f5ffb53e540 fr_remainder = 32608 retr_remainder = 473209280 retr_interval = 32765 new_retr_interval_ms = 547787427795 crt_retr_interval_ms = 140725076662752 t = 0x7f5ffb53e2a0 __func__ = "retr_buf_handler" #6 0x0000560e8a7375fa in slow_timer_main () at core/timer.c:1105 n = 12 ret = 0 tl = 0x7f5ffb53e560 i = 610 __func__ = "slow_timer_main"

2 2

Timeout for provisional response other that 100
by David Villasmil 09 Feb '21

09 Feb '21

Hello all, I’m trying to figure out if I can set. Timeout for specific provisional responses. I know I can set timeouts for INV, and restart the timer every time a provisional response is received. But this not what I actually need. What I need is to set a timeout from the 100 Trying to the next 180/183 (or a final response) and set back a 480 if not received. I’m not sure TM would allow me to do that, so maybe using a hash table to store transactions and calculate the time since the INV to the received 180/183. But there’s gotta be a simpler way. Many thanks! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Best SIP reply for service activation
by Duarte Rocha 09 Feb '21

09 Feb '21

Hello, I'm currently developing a service on Kamailio where it's possible to configure redirections for a subscriber using a code. In order to implement that client should send an INVITE with a specific code on the Request URI and the redirection is configured. What should be the best way to answer this INVITE? A 200 OK will make the call answered and doesn't seem the best way. A negative reply (4/5/6xx) would work, but it seems strange to have the operation concluded with success and answering with a negative reply. I found "199 - Early Dialog Terminated" to be suitable, but the documentation suggests it should be used upstream. Thanks in advance for any suggestion. Cheers

2 1

Kamailio inbound proxy -> Asterisk
by Mihai 09 Feb '21

09 Feb '21

Hi! Don't know if this is a Kamailio Question, maybe someone had deal with it in the past. I use Kamailio with dispatcher to send the request (registration, invites, all) to my asterisk (inbound) and then the asterisk has the outbound trunks, everything works, but this is little thing in asterisk when entering the "sip show peers" it shows me that all the peers are registered with the kamailio IP, if I set the nat option to no, it will show me the correct IP but breaks other clients. Any idea what to do ?

1 0

How to drop 200 OK in onreply_route ?
by Denys Pozniak 09 Feb '21

09 Feb '21

Hello! In the configuration below Kamailio Proxy creates a transaction for the SIP PUBLISH to get info from the HTTP server in async mode. But before creating a transaction, a synthetic 200 OK is sent, so that I need somehow to drop the real 200 OK from the upstream Presence Server. If drop 200 OK in *reply_route*, tm module starts to retransmit. Those it is necessary that the 200 OK be consumed by the tm module, but does not go further. *if ( !is_method("PUBLISH") ) {* * sl_send_reply("200", "Ok"); t_newtran();* * $http_req(suspend) = 1; http_async_query("$var(url)", "CALLBACK");* *}* *route[CALLBACK] {* * <some header manipulation>* * t_on_reply("PUBLISH_REPLY"); route(RELAY); exit;* *}* *onreply_route[PUBLISH_REPLY] { if ( t_check_status("200") ) { drop; # Does not work!!! }}* Any advice is appreciated -- BR, Denys Pozniak

3 8

Kamailio config file for making Kamailio a Sip Proxy
by Derick Greenlee 08 Feb '21

08 Feb '21

Hello, I had some questions regarding the Kamailio config file. I am trying to make Kamailio just behave as a sip proxy between two sip trunks. Could you send me an example for this possibly?

2 1

authenticated INVITE badly managed when topos is activated
by frédéric Gaisnon 08 Feb '21

08 Feb '21

Hi, If topos is activated and kamailio responds with a 407 "proxy authentication required" to an initial INVITE, INVITE received after with credentials (MD5 response in Proxy-Authorization header) is badly managed. Credentials are never validated and Kamailio responds 407 again. If topoh is used instead topos all works fine and credentials are validated. call flow is: Phone sends INVITE to kamailio SBC kamailio respond 407 "proxy authentication required" because SBC wants to authenticate caller Phone resends INVITE with Proxy-Authorization header with all valid information kamailio responds 407 again instead forwarding INVITE. My code: if (!pv_auth_check("$fd", "$sht(auth_cache=>$var(key))", "0", "1")) { auth_challenge("$fd", “1”); exit; } # user authenticated - remove auth header consume_credentials(); So if topos is used, pv_auth_check always returns false even if the phone sets valid information. If topoh is activated, pv_auth_check always returns true (same phone and same kamailio SBC versioning) I made my tests with kamailio 5.4.3 on centos 7. Regards, Frédéric Gaisnon

1 0

Registered with sip user using user pass
by Gaurav Bmotra 08 Feb '21

08 Feb '21

Hi I m using kamailio 5.3 OS Ubuntu 18 I wan to Register on sip server and then send traffic to to it form kamailio. Like : Client ---(sip peer)-->kamailio------(sip peer)---->gateway Please help me Thank you Regards Gaurav Kumar

1 0

filter_body("application/sdp") on multipart/mixed; boundary with double Content-Length
by Karsten Horsmann 08 Feb '21

08 Feb '21

Hi List, I have some weird multipart/mixed;boundary issues. I use kamailio 5.3.6 with rtpengine 7.x.x (same with 8.x.x). If i try to remove a multipart/mixed and leave only the sdp with filter_body() that works except one thing. The sender used IMHO an non RFC compliant version of rfc5621 and wrote the Content-Length Header twice (one in the header, one in the sdp). After manipulating the SDP filter_body() lets Content-Length in the sdp (i guess). And rtpengine is not happy with the sdp after that. rtpengine [rtpengine.c:2588]: rtpp_function_call(): proxy replied with error: Failed to parse SDP Any ideas how to solve this? Kind regards Karsten Horsmann https://tools.ietf.org/html/rfc5621#section-3.1 route[REMOVE_MULTIPART] { if (has_body("multipart/mixed")) { if ($sel(cfg_get.remove.multipart) == 1) { if (filter_body("application/sdp")) { remove_hf("Content-Type"); append_hf("Content-Type: application/sdp\r\n"); msg_apply_changes(); xlog("L_WARN", "[$cfg(route)] Body part multipart/mixed with application/sdp found\n"); } else { xlog("L_ERROR", "[$cfg(route)] Body part multipart/mixed without application/sdp found\n"); } } else { xlog("L_INFO", "[$cfg(route)] Body part multipart/mixed found - manipulation disabled\n"); } } } ------------------- INVITE sip:+49XXXXXXXX@172.20.120.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.120.51:5061;branch=z9hG4bK-6589-1-0 From: sipp <sip:sipp@172.20.120.51:5061>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXXX@172.20.120.57:5060> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Contact: sip:sipp@172.20.120.51:5061 Max-Forwards: 70 Subject: Performance Test Content-Type: multipart/mixed;boundary=sonus-content-delim Content-Length: 1185 --sonus-content-delim Content-Type: application/sdp Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 --sonus-content-delim Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicLoc" mlns:btd="http://btd.orange-business.com" entity=" pres:geotarget@btip.orange-business.com"><tuple id="XXXX"><status><gp:geopriv><gp:location-info><cl:civicAddress><cl:country>DE</cl:countr <cl:A1>Some State</cl:A1><cl:A3>Somewhere</cl:A3><cl:A4>Somewhere</cl:A4><cl:A6>Somestreet</cl:A6><cl:YYY>1</cl:YYY><cl:PC>9999</cl:PC><cl:EMERGENCYZONE>XXXXXXX</cl:EMERG CYZONE></cl:civicAddress></gp:location-info><gp:usage-rules></gp:usage-rules></gp:geopriv></status></tuple></presence> --sonus-content-delim-- --------------------- INVITE sip:+49XXXXXXXX@foo SIP/2.0 Via: SIP/2.0/TCP 212.XX.XX.XX;branch=z9hG4bKc228.6b8fdf20d79998d1d355feb2dcf5ba63.0 From: sipp <sip:sipp@foo>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXX@foo> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Max-Forwards: 69 Content-Length: 347 Content-Type: application/sdp P-Asserted-Identity: <sip:+49XXXXXXXX@212.XX.XX.XX> Contact: <sip:btpsh-60211bbe-674b-1@212.XX.XX.XX;transport=tcp> Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 -- Mit freundlichen Grüßen *Karsten Horsmann*

1 0

Retransmission of 407
by Marat Gareev 08 Feb '21

08 Feb '21

Hello, UAC sends INVITE to UAS, authorization procedure is required, therefore UAS responds with an 407 error code. At this point, the client does not respond with ACK, but the server does not send retransmits. Is this the expected behavior? * UAC -- SIPp, UAS -- Kamailio. Can the Kamailio be configured to send 407 code retransmissions? Thanks!

3 4

TO Header corrupted in RE-INVITE
by alirsm 07 Feb '21

07 Feb '21

Hi, Call to a number forwarded to a FAX by using two Kamailio proxies. (INVITE) --> KamailioA --> GW1 --> KamailioB --> GW2 --> FAX FAX --> (RE-INVITE) --> GW2 --> KamailioB --> (TO Header being corrupted) KamailioB is corrupting TO header when receiving RE-INVITE and causing the call to fail. 1) Only one "uac_replace_from()" is being used in each Kamailio. 2) GW1 is adding +1 (using E164 format) to URI and FROM headers. I was wondering if any of the above might cause the TO header corruption. Any help would be appreciated. AL

2 1

Failover with Dispatcher Module
by alirsm 05 Feb '21

05 Feb '21

Hi I have setup dispatcher failover on Kamailio 5.1 as below. If destination X.X.X.X is down, traffic should be routed to Y.Y.Y.Y. The failover is working but every time after 30 sec Timeout. Is there a way to setup health check properly so the failed destination can be ignored while it is down? Your help is greatly appreciated, AL # dispatcher.list 1 sip:X.X.X.X:5060 0 0 1 sip:Y.Y.Y.Y:5060 0 0 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_inactive_threshold", 4) modparam("dispatcher", "ds_ping_latency_stats", 1) modparam("dispatcher", "ds_probing_threshold", 1) # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("503", "No destination"); exit; } t_on_failure("DISPATCH_FAILOVER"); route(RELAY); exit; } # Failure Route failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if(!ds_next_dst()) { send_reply("503", "Out of gateways"); exit; } t_on_failure("DISPATCHER_FAILOVER"); t_relay(); }

2 1

Make configuration as proxy, push, renew server
by James Chan 02 Feb '21

02 Feb '21

Hi We use a Cloud-based SIP-server that does not support the Push-Notifications Service. We try to use Kamailio as the front-end to work as a proxy and push-server. We need help configuring Kamailio as a proxy and push-server. In addition, we want to implement auto renew configuration as UAC using Kamailio. Because all clients need to renew registration our Cloud-SIP-server every 59 minutes. ( registration expires after 60 minutes) Summary: UAC -> Kamailio (proxy & push & auto renew) -> Cloud-SIP-Server -> UAC I have attached our config file and screens And if phone has multiple contacts, there is call bug, call will be drop since there are two incoming call #!KAMAILIO #!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ACCDB #james #!define WITH_UAC #!define WITH_TSILO # # Kamailio (OpenSER) SIP Server v5.3 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!ifdef WITH_TSILO #!define FLB_TSILO 8 #!endif ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif fork=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:185.224.139.45:5060 /* 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 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* 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 = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_UAC loadmodule "uac.so" #!endif #!ifdef WITH_TSILO loadmodule "tsilo.so" loadmodule "htable.so" loadmodule "cfgutils.so" loadmodule "http_client.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_UAC modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 1) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif #!ifdef WITH_UAC modparam("uac", "reg_contact_addr", "pn.telesonique.ch:5060;transport=udp") modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_db_table", "uacreg") #!endif #!ifdef WITH_TSILO modparam("htable", "htable", "vtp=>size=10;autoexpire=180") modparam("http_client", "httpcon", "pushapi=>https://www.myswissline.com") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if (is_method("REGISTER")) { save("location", "0x06"); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } #james set_forward_no_connect(); if (!t_relay()) { sl_reply_error(); } #!ifdef WITH_TSILO else if(isflagset(FLB_TSILO)) { resetflag(FLB_TSILO); if (ts_store()) { $sht(tsilo=>stored::$rU) = 1; xdbg("stored transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route(PUSH_NOTIF); } #!endif exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies #set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } save("location", "0x06"); switch($rc) { case -1: case -2: sl_reply_error(); break; #!ifdef WITH_TSILO case 1: $var(new_contact) = 1; #!endif } #!ifdef WITH_TSILO if ($var(new_contact)) route(WAKE_SUSPENDED); #!endif exit; } #!ifdef WITH_TSILO route[WAKE_SUSPENDED] { $var(suspended) = 0; lock("$tU"); $var(suspended) = $sht(tsilo=>suspended::$tU); $sht(tsilo=>suspended::$tU) = $null; $var(stored) = $sht(tsilo=>stored::$tU); unlock("$tU"); if ($var(suspended) == 0) { if ($var(stored)) { # no suspended INVITE (i.e. there was at least one # registered contact for target) and 'stored' set # (i.e. ongoing transaction) -> this newly just # registered contact needs alerting. ts_append("location", "$tu"); } return; } $var(id_index) = $(var(suspended){s.select,0,:}{s.int}); $var(id_label) = $(var(suspended){s.select,1,:}{s.int}); xdbg("resuming trasaction [$var(id_index):$var(id_label)] $tU .\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); } route[RESUME] { if(! lookup("location")) { xlog("failed to load just added destinations\n"); exit; } setflag(FLB_TSILO); route(RELAY); } #!endif # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; #!ifdef WITH_TSILO send_reply("100", "Trying"); route(SUSPEND); route(PUSH_NOTIF); #!else route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } #!endif } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } #!ifdef WITH_TSILO setflag(FLB_TSILO); #!endif route(RELAY); exit; } #!ifdef WITH_TSILO route[SUSPEND] { if(! t_suspend()) { xlog("SUSPEND: failed to suspend transaction.\n"); # TODO: actually, the destination is not really checked: don't know if it's # own subscriber or invalid dialing in the first place. send_reply("503", "Temporarily unavailable"); exit; } $sht(tsilo=>suspended::$rU) = $_s($T(id_index):$T(id_label)); xdbg("suspended transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route[PUSH_NOTIF] { xdbg("TODO: send push for $rU\n"); $var(res) = http_connect("pushapi", "/send_push.php?username=$tU&dev=1", "application/json", "", "$avp(gurka)"); xlog("L_INFO", "Push notification: $avp(gurka) Result code $var(res)\n"); if ($var(res)==200) { if ($var(gurka)=~"^.*rejected.*$") { sl_send_reply("503", "Not available right now"); } else { sl_send_reply("101", "Push Sent"); } } if ($var(res)==28) { xlog("L_INFO", "Push notification timed out. Do we care?"); } exit; } #!endif # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_TSILO return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } [image: call_issue.png][image: register.png]

1 0

failover on SIP servers
by AL RSM 02 Feb '21

02 Feb '21

Hi, I was wondering what is the recommended approach to implement a failover on several destination SIP servers: 1) using DNS SRV records 2) using dispatcher list 3) combination of both. Thanks, AL

3 2

Acc module unable to insert cdr in mysql server
by M Arqum CH 01 Feb '21

01 Feb '21

Hello Members, Kamailio acc module is not inserting cdr into MySQL DB, sometimes it works and sometimes didn't, the number of parameters/values are similar in failed as well success scenario. I also check with tcpdump for MySQL requests, unfortunately, Kamailio didn't issue insert query toward DB server, so it is clear to me that the MySQL server is nothing to do with failed queries. Please guide me on what I should do to find and fix the issue. Kamailio Installed version : kamailio -v version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:691]: tmcb_func(): acc callback called for t(0x7fe29b2a7870) event type 512, reply code 200 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:413]: should_acc_reply(): probing acc state - code: 200 flags: 0x6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:446]: should_acc_reply(): acc is on Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 7 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 10 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 13 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 16 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 17 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 19 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 22 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 26 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 28 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 29 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 32 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 33 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_hooks.c:256]: run_trans_callbacks_internal(): DBG: trans=0x7fe29b2a7870, callback type 512, id 0 entered Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:267]: db_write_cdr(): failed to insert into database Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:682]: cdr_on_end_confirmed(): failed to write cdr! Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_reply.c:1758]: cleanup_uac_timers(): RETR/FR timers reset Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103682]: DEBUG: tm [timer.c:634]: wait_handler(): finished transaction: 0x7fe29727ba88 (p:0x7fe294d27288/n:0x7fe294d27288) -- Regards M Arqum

2 2

UACREG to many Asterisk
by Javier Valencia 01 Feb '21

01 Feb '21

Hi there! Go ahead I apologize for my English. I have a Kamailio 5.4 with the DISPATCHER module and UAC enabled (for remote REGISTER), the DISPATCHER has several SETIDs, and they have one or more Asterisk balanced behind it. In Kamalio script, when an endpoint does REGISTER, I enable the REGISTER of the UAC module. And when it expires or expires=0 I disable the REGISTER in the UAC module. In the UACREG table in the database, L_UUID (unique key), L_USERNAME, R_USERNAME, and AUTH_USERNAME have the same value. For example CST100EXT100. Knowing that the DISPATCHER and UAC modules have no relationship, but that when a SETID has several balanced Asterisks, how can I make the UAC module send the REGISTER to all the Asterisks of a SETID? Clarifications; 1.- I cannot update the Asterisk, which are currently in version 11. They do not support PATH. 2.- I can modify the content of the UACREG table so that l_uuid has a suffix '_X'. 3.- Call jsonrpc_exec several times to enable the REGISTER in the UAC module. (diff emulated) < jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_1"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_2"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_3"]}') ; But I want to know if there is any more elegant way to do this. Cheers, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

3 2

externally triggered message sending to specific clients
by Syed Shahryar 01 Feb '21

01 Feb '21

Is it possible to send arbitrary text messages triggered by an external application (evapi/http) to registered SIP clients via websocket? The following doesn't seem to do anything. KSR.corex.send_data("sip:user@example.com", "Message at $Ts"); -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

about kamailio-sbc-teams tutorial
by Juha Heinanen 01 Feb '21

01 Feb '21

I read kamailio-sbc-teams tutorial https://skalatan.de/blog/kamailio-sbc-teams and it does not mention, which CA root certificates need to be included tls config ca_list so that Kamalio is able to verify sipX.pstnhub.microsoft.com server certificates. -- Juha

2 1

logging error events to database
by Sergiu Pojoga 30 Jan '21

30 Jan '21

Hi there, I was wondering if there's a way to log various core error events in some DB. For example, below error occurred Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE 28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core> [core/parser/sdp/sdp_helpr_funcs.c:499]: extract_mediaip(): no `IP[4|6]' address in `c=' field Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE 28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core> [core/parser/sdp/sdp.c:430]: parse_sdp_session(): can't extract common media IP from the message Thanks in advance. Regards, --Sergiu

4 5

Establish connection between MsTeams and Kamailio.
by Willy Valles Rios 30 Jan '21

30 Jan '21

Hello community, Please your support, I tell you that I am integrating Kamailio with MsTeams, after some time reviewing this, I finally achieved the connection from my Kamailio server to MsTeams and I can validate that the connection to MsTeams is in AP. [root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI | FLAGS" URI: sip: sip.pstnhub.microsoft.com; transport = tls FLAGS: AP However, from the admin panel of MsTeams (Direct Routing) I see that the connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity status in "Active" but the SIP options status is in "Warning". I have made calls from MsTeams thinking that the SIP options status would change to "active" but it is still in "Warning" state. On the other hand, I have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from kamailio are being sent in the following format to MsTeams. OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0 Via: SIP / 2.0 / TLS 161.35.44.66:5061 ;branch=z9hG4bKea07.52224687000000000000000000000000.0 To: <sip: sip.pstnhub.microsoft.com; transport = tls> From: <sip: sbc.netvoiceperu.com>; tag = d3569c818b500aeb8c373426e76c2884-81763c71 CSeq: 10 OPTIONS Call-ID: 13ea237a751e0c48-9148(a)161.35.44.66 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.0 (x86_64 / linux)) As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not contain the "Contact" field, which in theory said "Contact" field should have been added by Kamailio according to the configuration added in kamailio.cfg event_route [tm: local-request] { sip_trace (); if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; transport = tls> \ r \ n"); } xlog ("L_INFO", "Sent out tm request: $ mb \ n"); } As additional information, I inform you that I also managed to observe the SIP OPTIONS that MsTeams sends to Kamailio. OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0 FROM: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>; tag = f1bdeb5f-662f-4544-a436-e9aa9ad78da4 TO: <sip: sbc.netvoiceperu.com> CSEQ: 1 OPTIONS CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a MAX-FORWARDS: 70 VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be CONTACT: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10 ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY However I don't see the 200 OK SIP responses from Kamailio to MsTeams. I think this may be the reason why I see the SIP OPTIONS status in "Warning" from the MsTeams panel. Maybe the contact field is not being added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for that reason I don't see 200OK responses from MsTeams. Could you help me solve this please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com

2 1

registering gsm gateway by siremis
by shashank shekhar 29 Jan '21

29 Jan '21

Hi, I want to register a gsm gateway on my kamailio server with siremis ...kindly guide me on this Thankyou Regards Shashank shekhar 9162735161

2 1

Kamailio as inbound and outbound proxy
by Michael Friesen 29 Jan '21

29 Jan '21

Hello, I am trying to configure Kamailio as both an inbound and outbound proxy. Inbound requests flow Sip Trunk -> Kamailio -> Asterisk. Outbound requests flow Asterisk -> Kamailio -> Sip Trunk. Inbound traffic is sent to port 5060 on kamailio which listens on the PRIVATE_IP:5060 and advertises PUBLIC_IP:5060. How can we ensure that when messages are sent to the Sip Trunk they have Record-Route headers with the PUBLIC_IP and when messages are sent to asterisk they have Record-Route headers with the PRIVATE_IP?

2 1

kamailio 5.4.3 ubuntu 20.04 tls - http_async_client
by Filippo Graziola 28 Jan '21

28 Jan '21

Hi all, I have an issue related (my guess) to tls and http_async_client module that result in a segmentation fault and a not correct handle of tls connections. First with only tls module loaded, not forked: 0(1021) INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1021) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1021) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1021) INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! 0(1021) INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman 0(1021) WARNING: tls [tls_init.c:784]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 4718592 and 2359296 bytes 0(1021) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 4718592 0(1021) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2359296 0(1021) INFO: <core> [main.c:2833]: main(): processes (at least): 9 - shm size: 67108864 - pkg size: 67108864 0(1021) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1021) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1021) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 0(1021) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/fullchain.pem' 0(1021) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' 0(1021) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' 0(1021) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 0(1021) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' 0(1021) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/privkey.pem' 0(1021) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 0(1021) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 0(1021) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 0(1021) NOTICE: tls [tls_domain.c:1105]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... 0(1021) INFO: tls [tls_domain.c:711]: set_verification(): TLSs<default>: No client certificate required and no checks performed 0(1021) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 0(1021) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' 0(1021) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' 0(1021) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' 0(1021) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 0(1021) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' 0(1021) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' 0(1021) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 0(1021) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 0(1021) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 0(1021) INFO: tls [tls_domain.c:714]: set_verification(): TLSc<default>: Server MAY present invalid certificate 6(1027) ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error 6(1027) ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:141FC044:SSL routines:tls_setup_handshake:internal error 6(1027) ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: XXXXXXXXXXXXXXX 6(1027) ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: XXXXXXXXXX 6(1027) ERROR: <core> [core/tcp_read.c:1498]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2cbc1b3948 r: 0x7f2cbc1b3a70 (-1) so no segmentation fault but error in handling. Second one also with http_async_client loaded: 0(1059) INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1061) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1061) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1061) INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! 0(1061) INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman 0(1061) INFO: http_async_client [http_async_client_mod.c:222]: mod_init(): Initializing Http Async module 0(1061) WARNING: tls [tls_init.c:784]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(1061) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(1061) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(1061) INFO: <core> [main.c:2833]: main(): processes (at least): 10 - shm size: 67108864 - pkg size: 67108864 0(1061) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1061) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1061) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 0(1061) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/fullchain.pem' 0(1061) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' 0(1061) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' 0(1061) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 0(1061) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' 0(1061) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/privkey.pem' 0(1061) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 0(1061) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 0(1061) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 0(1061) NOTICE: tls [tls_domain.c:1105]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... 0(1061) INFO: tls [tls_domain.c:711]: set_verification(): TLSs<default>: No client certificate required and no checks performed 0(1061) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 0(1061) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' 0(1061) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' 0(1061) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' 0(1061) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 0(1061) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' 0(1061) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' 0(1061) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 0(1061) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 0(1061) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 0(1061) INFO: tls [tls_domain.c:714]: set_verification(): TLSc<default>: Server MAY present invalid certificate 0(1061) INFO: http_async_client [async_http.c:101]: async_http_init_sockets(): inter-process event notification sockets initialized 0(1061) INFO: http_async_client [async_http.c:84]: async_http_init_worker(): started worker process: 1 0(1059) CRITICAL: <core> [core/mem/q_malloc.c:501]: qm_free(): BUG: bad pointer 0x1 (out of memory block!) called from tls: tls_init.c: ser_free(323) - ignoring Segmentation fault this time, there is a segmentation fault. The above is a result of this minimal configuration: #!KAMAILIO ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " children=2 tcp_children=2 auto_aliases=no alias="XXXXXXXXXXXXX" listen=udp:eth0 server_signature=no tcp_connection_lifetime=3605 tcp_max_connections=40960 tcp_accept_no_cl=yes enable_tls=yes listen=tls:XXXXXXXXXX:5061 advertise XXXXXXXXXXXX:5061 tls_max_connections=40000 enable_sctp=no ####### Modules Section ######## loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "tls.so" loadmodule "http_async_client.so" #----------------- setting module-specific parameters --------------- #----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #----- http client ---- modparam("http_async_client", "workers", 1) ####### Routing Logic ######## request_route { exit; } I used the above configuration to take out as much as possible my mistakes in the configuration, but with my full kamailio configuration, tls connections give the above errors but everything else works just fine (also http_async_client module functions which are used on INVITES) and calls are going properly (unfortunately tls is required). I found a couple of issues that are similar https://github.com/kamailio/kamailio/issues/2560 and https://github.com/kamailio/kamailio/issues/2466# but as far as I understood the issue 2466 is closed because fixes are already included. I tried in any case to compile from source a few older releases with the same result, changed also the certificate and private key (letsencrypt), moreover I have another kamailio (v5.3.4) running on ubuntu 18.04 (same configuration) without any issues. I saw that there is a different version of openssl version 1.0.. in ubuntu 18.04, version 1.1 in ubuntu 20.04, but the segmentation fault seems to happen after an error on free some memory. Have you some ideas? tell me if you need more info from me. Thanks Filippo

4 7

Problems establishing SIP signaling between MsTeams and Kamailio
by Willy Valles Rios 28 Jan '21

28 Jan '21

Hello community, I am having trouble establishing SIP signaling between MsTeams and Kamailio. I currently have this configuration in my tls.cfg file [server: default] method = TLSv1.2 + verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certificates/private-key.pem certificate = /etc/kamailio/certificates/certificate.pem [client: default] method = TLSv1.2 + verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certificates/private-key.pem certificate = /etc/kamailio/certificates/certificate.pem My domain was certified with ssl through an authoritative certifier (GoDaddy), however I see these errors in the / var / log / messages of the Kamailio server. Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_init.c:722]: tls_h_mod_init_f(): compiled with openssl version "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, compression: on Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_init.c:730]: tls_h_mod_init_f(): installed openssl library version "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, zlib compression: on#012 compiler: gcc -I. -I.. -I../include -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic -Wa,--noexecstack -DPURIFY -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM -DECP_NISTZ256_ASM Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: WARNING: tls [tls_init.c:787]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 13107200 and 6553600 bytes Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 13107200 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 6553600 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [main.c:2834]: main(): processes (at least): 25 - shm size: 67108864 - pkg size: 4194304 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs: tls_method=22 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs: certificate='/etc/kamailio/certificados/certificate.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs: ca_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs: crl='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:335]: ksr_tls_fill_missing(): TLSs: require_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs: cipher_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs: private_key='/etc/kamailio/certificados/private-key.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:353]: ksr_tls_fill_missing(): TLSs: verify_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs: verify_depth=9 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs: verify_client=0 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: NOTICE: tls [tls_domain.c:1107]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='' ... Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:697]: set_verification(): TLSs: Client MUST present valid certificate Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc: tls_method=22 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc: certificate='/etc/kamailio/certificados/certificate.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc: ca_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc: crl='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:335]: ksr_tls_fill_missing(): TLSc: require_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc: cipher_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc: private_key='/etc/kamailio/certificados/private-key.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:353]: ksr_tls_fill_missing(): TLSc: verify_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc: verify_depth=9 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc: verify_client=0 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:697]: set_verification(): TLSc: Server MUST present valid certificate Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32422]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/32422 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32424]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.75.24 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242be028 r: 0x7f45242be150 (-1) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.132.46 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242d9278 r: 0x7f45242d93a0 (-1) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.14.70 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242be028 r: 0x7f45242be150 (-1) Could you help me identify the problem please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com

9 18

Info: Upcoming online events - February 2021
by Daniel-Constantin Mierla 27 Jan '21

27 Jan '21

Hello, short note to announce a news article listing the upcoming online events where you can watch presentation related to Kamailio, respectively FOSDEM and ClueCon TGI. More details at: * https://www.kamailio.org/w/2021/01/upcoming-online-events-february-2021/ During this time where the travelling is under heavy restrictions, such events can be a chance to keep in touch with peers in the RTC space or learn new usefuk things. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Loop with unable to find dialog for xxx
by David VILLAUME 27 Jan '21

27 Jan '21

Hello, I got some time Ack packets looping on my servers during 4 hours if I don’t restart the system ( 4 hours is my dialog timeout). Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' It generates approximatively 2 millions packets per loop. On the last occurrence I noticed a ‘SIP/2.0 200 canceling’ generated per Kamailio before the loop started (which seems uncomon on this host) Bellow is a screenshot of the beginning of the loop and may dialog and topos config. Has anyone encountered such behavour ? Regards, David [cid:image001.png@01D6F4E2.F68B5140] modparam("dialog", "dlg_flag", 1) modparam("dialog", "hash_size", 1024) modparam("dialog", "default_timeout", 14500) modparam("dialog", "event_callback", "ksr_dialog_event") modparam("dialog", "db_url", DBURL_DLG) modparam("dialog", "db_update_period", 15) modparam("dialog", "db_mode", 1) modparam("topos", "storage", "redis") modparam("topos", "dialog_expire", 14500) modparam("topos", "contact_host", "PUBLICIP") modparam("topos_redis", "serverid", "topos")

1 0

websocket rd buffer size limit?
by Andrew Chen 27 Jan '21

27 Jan '21

Hey group, I was wondering if there is a way to control the websocket buffer size to support large SIP messages. I received the following error when sending content-length of 65000 2021-01-26T17:32:19.251408+00:00 sjomainkama50 kamailio[31871]: 109(32008) WARNING: websocket [*ws_frame.c*:482]: decode_and_validate_ws_frame(): message is too long for our buffer size (65535 / 66151) I am assuming this is controlled by the tcp_rd_buf params which I have set to 131k: *tcp_*rd_buf_size=131072 and this is my tcp_wr_buf params: *tcp_*wq_blk_size=131072 *tcp_*conn_wq_max=131072 Please advise. Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 4

event_route[core:receive-parse-error] missing from docs
by Henning Westerholt 27 Jan '21

27 Jan '21

Hello, I noticed that the core "event_route[core:receive-parse-error]" was missing from the cookbook, so I have added it there. https://www.kamailio.org/wiki/cookbooks/devel/core#event_route Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

FQDN endpoint as rd
by AL RSM 27 Jan '21

27 Jan '21

Hi, I was wondering if Kamailio can support FQDN endpoint which resolves to multiple IPs, checks if the first IP is not reachable, try the next one in the list. Thanks, AL

2 1

Re: [SR-Users] sr-users Digest, Vol 188, Issue 9
by Willy Valles Rios 26 Jan '21

26 Jan '21

Hello community, After a while, I finally got the connection from my Kamailio server to MsTeams and I can validate that the connection to MsTeams is in AP. [root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI | FLAGS" URI: sip: sip.pstnhub.microsoft.com; transport = tls FLAGS: AP However, from the admin panel of MsTeams (Direct Routing) I see that the connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity status in "Active" but the SIP options status is in "Warning". I have made calls from MsTeams thinking that the SIP options status would change to "active" but it is still in "Warning" state. On the other hand, I have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from kamailio are being sent in the following format to MsTeams. OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0 Via: SIP / 2.0 / TLS 161.35.44.66:5061 ;branch=z9hG4bKea07.52224687000000000000000000000000.0 To: <sip: sip.pstnhub.microsoft.com; transport = tls> From: <sip: sbc.netvoiceperu.com>; tag = d3569c818b500aeb8c373426e76c2884-81763c71 CSeq: 10 OPTIONS Call-ID: 13ea237a751e0c48-9148(a)161.35.44.66 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.0 (x86_64 / linux)) As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not contain the "Contact" field, which in theory said "Contact" field should have been added by Kamailio according to the configuration added in kamailio.cfg event_route [tm: local-request] { sip_trace (); if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; transport = tls> \ r \ n"); } xlog ("L_INFO", "Sent out tm request: $ mb \ n"); } As additional information, I inform you that I also managed to observe the SIP OPTIONS that MsTeams sends to Kamailio. OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0 FROM: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>; tag = f1bdeb5f-662f-4544-a436-e9aa9ad78da4 TO: <sip: sbc.netvoiceperu.com> CSEQ: 1 OPTIONS CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a MAX-FORWARDS: 70 VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be CONTACT: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10 ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY However I don't see the 200 OK SIP responses from Kamailio to MsTeams. I think this may be the reason why I see the SIP OPTIONS status in "Warning" from the MsTeams panel. Maybe the contact field is not being added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for that reason I don't see 200OK responses from MsTeams. Could you help me solve this please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com El dom, 10 ene 2021 a las 6:03, <sr-users-request(a)lists.kamailio.org> escribió: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Problems establishing SIP signaling between MsTeams and > Kamailio (Carlos Mestanza T.) > 2. Re: Problems establishing SIP signaling between MsTeams and > Kamailio (Rob van den Bulk) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 8 Jan 2021 19:16:30 -0500 > From: "Carlos Mestanza T." <mestacart(a)gmail.com> > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, "Daniel Constantin Mierla > (asipto)" > <miconda(a)gmail.com> > Subject: Re: [SR-Users] Problems establishing SIP signaling between > MsTeams and Kamailio > Message-ID: > < > CAHcXM7PN0zuJC9jONTERZdfur3b+ucwnCC712Ss1OBxCiMctUg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Dear Kamilio Community, > > After doing several reads to our Kamailio configuration, the dispatcher > list is in AP: > > URI: sip: sip.pstnhub.microsoft.com; transport = tls > FLAGS: AP > URI: sip: sip2.pstnhub.microsoft.com; transport = tls > FLAGS: AP > URI: sip: sip3.pstnhub.microsoft.com; transport = tls > FLAGS: AP > > But in the MS Teams dashboard the SIP OPTIONS STATUS column is WARNING. > > In the LOGs sent to OPTIONS, it gives us to understand that with sent to MS > TEAMS: > > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKd25f.2835f676000000000000000000000000.0#015#012To: > <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-213e3c71#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d0-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d0-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip2.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKe25f.b14dc514000000000000000000000000.0#015#012To: > <sip:sip2.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-44c3af70#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d1-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d1-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip3.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKb25f.8442f914000000000000000000000000.0#015#012To: > <sip:sip3.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-fa555adb#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d2-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d2-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:42 Kamailio-Server journal: Suppressed 103 messages from > /system.slice/kamailio.service > > In the kamailio.cfg configuration it is declared: > > listen=tls:161.35.44.66:5061 > listen=tcp:10.131.245.99:5061 > > modparam ("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") > modparam ("dispatcher", "ds_probing_mode", 1) > modparam ("dispatcher", "ds_ping_interval", 60) > > And this fragment was also added: > > event_route [tm: local-request] { > > if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { > append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; > transport = tls> \ r \ n"); > } > xlog ("L_INFO", "Sent out tm request: $ mb \ n"); > } > > There is something additional that has to be declared so that in the MS > Teams panel the SIP OPTIONS STATUS column is shown as ACTIVE and not as > WARNNING, in the MS TEAMS documentation it is a possible problem related to > OPTIONS events. > > Atentamente > > *Adalberto Carlos Mestanza T.* > > > > El jue, 7 ene 2021 a las 21:54, Ovidiu Sas (<osas(a)voipembedded.com>) > escribi?: > > > That certificate should already be present under the OS's trusted > > certificates directory (debian and ubuntu certs are stored under > > /etc/ssl/certs), maybe under a different name, and is required for > > remote endpoint's certificate validation. > > One can load a particular certificate or a list of certificates. > > Multiple certificates can be concatenated into one single file as > > stated in the documentation: > > https://kamailio.org/docs/modules/devel/modules/tls.html#tls.p.ca_list > > Hope this helps a little bit in understanding of the ca_list param. > > > > Regards, > > Ovidiu Sas > > > > On Thu, Jan 7, 2021 at 8:10 AM <rob.van.den.bulk(a)gmail.com> wrote: > > > > > > I Used this tls.cfg > > > > > > > > > > > > Use bc2025.pem as extra, Microsoft needs this? > > > > > > > > > > > > And works fine on different Kamailio-msteams sbcs > > > > > > > > > > > > > > > > > > [server:default] > > > > > > method = TLSv1.2+ > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/privkey.pem > > > > > > certificate = > > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/fullchain.pem > > > > > > ca_list = /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/bc2025.pem > > > > > > server_name = sbc.combivoipdom.nl > > > > > > > > > > > > [client:default] > > > > > > method = TLSv1.2+ > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/privkey.pem > > > > > > certificate = > > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/fullchain.pem > > > > > > ca_list = /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/bc2025.pem > > > > > > > > > > > > > > > > > > > > > > > > Cheers Rob > > > > > > > > > > > > Van: sr-users <sr-users-bounces(a)lists.kamailio.org> Namens > > Daniel-Constantin Mierla > > > Verzonden: donderdag 7 januari 2021 08:53 > > > Aan: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org > >; > > Willy Valles Rios <willyvalles17(a)gmail.com> > > > CC: Carlos Mestanza T. <mestacart(a)gmail.com> > > > Onderwerp: Re: [SR-Users] Problems establishing SIP signaling between > > MsTeams and Kamailio > > > > > > > > > > > > Does this happen when Kamailio connects to MS Teams? The logs indicate > > the received TLS certificate is not trusted: > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > > > > > > > You can set debug=3 in kamailio.cfg and see if the DEBUG messages > > provide more hints. For me it worked fine with Letsencrypt certs in > > Kamailio and accepting what ever MS sent back. I used Debian 10 and > libssl > > 1.1. > > > > > > > > > > > > Cheers, > > > Daniel > > > > > > > > > > > > On 06.01.21 21:47, Willy Valles Rios wrote: > > > > > > Hello community, > > > > > > > > > > > > I am having trouble establishing SIP signaling between MsTeams and > > Kamailio. I currently have this configuration in my tls.cfg file > > > > > > > > > > > > [server: default] > > > > > > method = TLSv1.2 + > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = /etc/kamailio/certificates/private-key.pem > > > > > > certificate = /etc/kamailio/certificates/certificate.pem > > > > > > > > > > > > [client: default] > > > > > > method = TLSv1.2 + > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = /etc/kamailio/certificates/private-key.pem > > > > > > certificate = /etc/kamailio/certificates/certificate.pem > > > > > > > > > > > > My domain was certified with ssl through an authoritative certifier > > (GoDaddy), however I see these errors in the / var / log / messages of > the > > Kamailio server. > > > > > > > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_mod.c:389]: mod_init(): With ECDH-Support! > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_mod.c:392]: mod_init(): With Diffie Hellman > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_init.c:722]: tls_h_mod_init_f(): compiled with openssl version > > "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, > > compression: on > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_init.c:730]: tls_h_mod_init_f(): installed openssl library version > > "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, > zlib > > compression: on#012 compiler: gcc -I. -I.. -I../include -fPIC > -DOPENSSL_PIC > > -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H > -DKRB5_MIT > > -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 > > -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 > > -grecord-gcc-switches -m64 -mtune=generic -Wa,--noexecstack -DPURIFY > > -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 > > -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM > > -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM > > -DECP_NISTZ256_ASM > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: WARNING: tls > > [tls_init.c:787]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks > on > > low memory) workaround enabled (on low memory tls operations will fail > > preemptively) with free memory thresholds 13107200 and 6553600 bytes > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold1 has been > > changed to 13107200 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold2 has been > > changed to 6553600 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [main.c:2834]: main(): processes (at least): 25 - shm size: 67108864 - > pkg > > size: 4194304 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is > initially > > 212992 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally > > 425984 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs: tls_method=22 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs: > > certificate='/etc/kamailio/certificados/certificate.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs: ca_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs: crl='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:335]: ksr_tls_fill_missing(): TLSs: require_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs: cipher_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs: > > private_key='/etc/kamailio/certificados/private-key.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:353]: ksr_tls_fill_missing(): TLSs: verify_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs: verify_depth=9 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs: verify_client=0 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: NOTICE: tls > > [tls_domain.c:1107]: ksr_tls_fix_domain(): registered server_name > callback > > handler for socket [:0], server_name='' ... > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:697]: set_verification(): TLSs: Client MUST present valid > > certificate > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc: tls_method=22 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc: > > certificate='/etc/kamailio/certificados/certificate.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc: ca_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc: crl='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:335]: ksr_tls_fill_missing(): TLSc: require_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc: cipher_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc: > > private_key='/etc/kamailio/certificados/private-key.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:353]: ksr_tls_fill_missing(): TLSc: verify_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc: verify_depth=9 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc: verify_client=0 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:697]: set_verification(): TLSc: Server MUST present valid > > certificate > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32422]: INFO: > jsonrpcs > > [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/32422 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32424]: INFO: ctl > > [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io > > watch method (config) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.75.24 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242be028 r: 0x7f45242be150 (-1) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.132.46 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242d9278 r: 0x7f45242d93a0 (-1) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.14.70 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242be028 r: 0x7f45242be150 (-1) > > > > > > > > > > > > Could you help me identify the problem please. > > > > > > > > > > > > Cheers > > > > > > > > > > > > Saludos Cordiales > > > > > > -- > > > > > > Willy Valles Rios > > > > > > Unified Communications Specialist > > > > > > > > > > > > phone: +51955747343 > > > > > > em@il: willyvalles17(a)gmail.com > > > > > > > > > > > > _______________________________________________ > > > > > > Kamailio (SER) - Users Mailing List > > > > > > sr-users(a)lists.kamailio.org > > > > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > > > > > > Daniel-Constantin Mierla -- www.asipto.com > > > > > > www.twitter.com/miconda -- www.linkedin.com/in/miconda > > > > > > Funding: https://www.paypal.me/dcmierla > > > > > > _______________________________________________ > > > Kamailio (SER) - Users Mailing List > > > sr-users(a)lists.kamailio.org > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > VoIP Embedded, Inc. > > http://www.voipembedded.com > > >

2 1

Install kamailio-tls-4.4.7-1.1.x86_64.rpm on centos 8
by Ahmed Marsou 25 Jan '21

25 Jan '21

Hi; There is any package ready to install kamailio-tls-4.4.7-1.1.x86_64.rpm on centos 8? Thank you.

2 1

Version 5.4 contact rewrite/restore functions docs?
by Dmitry Sytchev 22 Jan '21

22 Jan '21

Hello! In 5.4 release announce there is a line about contact rewriting. "new Contact address encoding/decoding functions – replace/restore Contact address with SIP server address, by encoding/decoding old address in a parameter" Where can I find some documentation or examples about NEW functions? At present we handle this task exactly the same way via parameters manually. It is kinda verbose and hard to understand. Maybe new functions will simplify this. Please advise! Thanks! -- Best regards, Dmitry Sytchev, IT Engineer

2 3

Help required in sipwise rtpengine-ng-client and rtpengine-ctl
by mahesh prasad behera 22 Jan '21

22 Jan '21

Hi Team, We are using sipwise rtpengine on platform centos 7. To get call related statistics from rtpengine. We tried to use utils "rtpengine-ng-client" and "rtpengine-ctl". But for us both of them are not working. *Rtpengine process status:* [root@ctl utils]# ps -ef |grep rtpengine root 3924 668 0 00:11 pts/0 00:00:00 grep --color=auto rtpengine root 20502 20499 0 Jan19 ? 00:10:02 ../sbin/rtpengine -f --num-threads 4 -i pub/10.211.160.132 -i priv/10.211.160.132 -n 127.0.0.1:8500 -c 127.0.0.1:8500 -m 32001 -M 32500 -T 184 -o 90 -d 4 -s 900 -p /var/run/rtpengine1.pid --scheduling rr --priority 37 [root@ctl utils]# * rtpengine-ng-client:* When i ran rtpengine-ng-client, I was getting Bencode.pm missing so I have manually installed perl bencode library "perl-Convert-Bencode-1.03-9.el7.noarch.rpm" Now when i ran rtpengine-ng-client i am getting below error [root@ctl utils]# ./rtpengine-ng-client list *Undefined subroutine &Bencode::bencode called at /usr/local/lib64/perl5/NGCP/Rtpengine.pm line 33. * *rtpengine-ctl:* When i ran ./rtpengine-ctl -ip 127.0.0.1:8500 list , We are not getting any valid response from rtpengine. [root@ctl utils]# ./rtpengine-ctl -ip 127.0.0.1:8500 list *Inside do while after call socket->recv(response, 1024*1024*10)* Please find the below attached source code for rtpengine-ng-client and rtpengine-ctl. Any help and suggestion will be highly appreciated. Thanks, Mahesh

2 1

Kemi execute a route from configuration file
by David VILLAUME 22 Jan '21

22 Jan '21

Hello, I was pretty sure there was a way to call a route from config file from Kemi, but I can’t find documentation about it. Does that really exists or did I dreamed it ? Regards, David

2 2

Kamailio, debian and MySQL caching_sha2_password
by Nicki Otte 22 Jan '21

22 Jan '21

Hi all Have any of you experienced problems using MySQL(Not MariaDB) on a debian machine, with caching_sha2_password? I am having some trouble: Jan 21 08:13:41 MobileValue-Kamailio systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Jan 21 08:13:42 MobileValue-Kamailio kamailio: INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: Listening on Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: udp: 185.31.176.39:5060 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: tcp: 185.31.176.39:80 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: Aliases: Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: tcp: voipgw01.evercall.dk:80 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: udp: voipgw01.evercall.dk:5060 Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Plugin caching_sha2_password could not be loaded: /usr/lib/x86_64-linux-gnu/mariadb19/plugin/caching_sha2_password.so: cannot open shared object file: No such file or directory Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatch.c:903]: ds_connect_db(): cannot initialize db connection Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatch.c:937]: ds_init_db(): unable to connect to the database Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatcher.c:404]: mod_init(): could not initiate a connect to the database Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module dispatcher (/usr/lib/x86_64-linux-gnu/kamailio/modules/dispatcher.so) Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: kamailio.service: Control process exited, code=exited, status=255/EXCEPTION Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: kamailio.service: Failed with result 'exit-code'. Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. I can get around the problem, by making the kamailio & kamailioro user use mysql_native_password instead. But am i wrong to belieave that it should Work out of the box? Here are the installed packages from APT: kamailio/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-json-modules/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-mysql-modules/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-outbound-modules/unknown,now 5.4.3+bpo10 amd64 [installed] default-libmysqlclient-dev/stable,now 1.0.5 amd64 [installed] default-libmysqld-dev/stable,now 1.0.5 amd64 [installed] libmariadb-dev/stable,now 1:10.3.27-0+deb10u1 amd64 [installed,automatic] libmariadb-dev-compat/stable,now 1:10.3.27-0+deb10u1 amd64 [installed,automatic] mysql-apt-config/unknown,now 0.8.16-1 all [installed] mysql-client/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-common/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client-core/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client-plugins/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-server/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-server-core/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-server/unknown,now 8.0.23-1debian10 amd64 [installed] Thanks in advance Nicki Bo Otte

4 8

RTP Querys
by Ahmed Marsou 22 Jan '21

22 Jan '21

Hi guys; I have some rtpproxy questions: - When a proxy is in a failed state, what happens to the active sessions / streams? - How many times the module should retry to send and receive after timeout was generated? - *How often are the control messages sent to the rtp proxy’s in the pool?* *- I would like to know the exact nc message that’s used for the control messages* *Thank you.*

2 2

tracking usrloc clean
by Michal Popovic 22 Jan '21

22 Jan '21

Hi All, I would like to ask you if there is any way how to track usrloc location cleaning operation with some event route, so I can broadcast the change with for example kafka message. If no, how do you keep track of online/offlice status of the user? I am trying to avoid using change data capture in mysql proxy for location table. Michal

2 2

Kamailio and Serverless
by H Yavari 22 Jan '21

22 Jan '21

Hi to all, Does anyone have experience using Kamailio in front of some Lambda / Cloud functions? For example, passing call info with JSON to functions.Please share your ideas. Regards,HYavari

4 4

Kamailio v5.4 install
by Jewna 21 Jan '21

21 Jan '21

I'm having a problem installing kamailio 5.4 using the following guide https://kamailio.org/docs/tutorials/5.4.x/kamailio-install-guide-git/ . I have a debian 10.7 machine where I install default-mysql-server package for the database. After that I follow the install guide and everything goes smoothly until I have to install the init.d scripts. 1. I do "cd /usr/local/src/kamailio-5.4/kamailio' 2. I do "make install-initd-debian" Then make fails with the following error " root@kamServ:/usr/local/src/kamailio-5.4/kamailio# make install-initd-debian make -C src/ install-initd-debian make[1]: Entering directory '/usr/local/src/kamailio-5.4/kamailio/src' sed -e "s#DAEMON=/usr/sbin/kamailio#DAEMON=/usr/local/sbin/kamailio#g" \ -e "s#NAME=kamailio#NAME=kamailio#g" \ -e "s#DESC=Kamailio#DESC=kamailio#g" \ -e "s#HOMEDIR=/run/kamailio#HOMEDIR=/run/kamailio#g" \ -e "s#DEFAULTS=/etc/default/kamailio#DEFAULTS=/etc/default/kamailio#g" \ -e "s#CFGFILE=/etc/kamailio/kamailio.cfg#CFGFILE=/usr/local/etc/kamailio/kamailio.cfg#g" \ < pkg/kamailio/deb/debian/kamailio.init \ > /etc/init.d/kamailio /bin/sh: 1: cannot open pkg/kamailio/deb/debian/kamailio.init: No such file make[1]: *** [Makefile:982: install-initd-debian] Error 2 make[1]: Leaving directory '/usr/local/src/kamailio-5.4/kamailio/src' make: *** [Makefile:34: install-initd-debian] Error 2 " Am I doing something wrong? Possibly in the wrong directory? If you need any more information please don't hesitate to ask. Sincerely, Matthew Sherwood

2 1

[kamailio 5.4.3] xavp_rm_by_index removes (N+1) first indexes instead of index N
by Chaigneau, Nicolas 20 Jan '21

20 Jan '21

Hello I’m trying to use function xavp_rm_by_index to remove a specific index from a given xavp. I’m observing an unexpected behaviour, I think it might be a bug : Instead of just removing the index, the function removes the index and all others before it. E.g : if called with « idx = 1 », it removes indexes 0 and 1. Reading the code of xavp_rm_internal (which is called by xavp_rm_by_index), I don’t think it behaves at it should : /* Remove xavps * idx: <0 remove all xavps with the same name * >=0 remove only the specified index xavp * Returns number of xavps that were deleted */ For example when I do this : LM_ERR("COUNT BEFORE: %d\n", xavp_count(rname, NULL)); int rm_count = xavp_rm_by_index(rname, 1, NULL); // TEST LM_ERR("REMOVED: %d\n", rm_count); LM_ERR("COUNT AFTER: %d\n", xavp_count(rname, NULL)); « rm_count » returned is 1 (which is expected). However, « count after » is decremented by 2. And when I look at the contents of my xavp, I notice that the first two elements have been removed. Likewise if xavp_rm_by_index is invoked with idx = 2 => the first 3 elements are removed. So… do you think this is a bug ? or something I didn’t understand ? Thanks for your help ! Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

Dialog Tracking with Inconsistent RR
by Charles Phillips 20 Jan '21

20 Jan '21

I am working to track concurrent channels with the dialog module, but I have run in to an issue. Due to the nature of my required configuration, I have a need to strip the record route headers from packets that are sent to the upstream proxy. This leaves no way for the dialog module to track the state of the call. I was hoping that DID_FALLBACK or DID_NONE modes would track by call id or some other constant, but it does not appear to. So, before I start down the path of using htable to store and reinsert the did when a packet returns, is there something I am missing with dialog? Is there some more efficient way to do this? - Charles

2 2

Error: invalid json doc in command "kamctl ping" kamailio 5.4.3
by Julia Bo 20 Jan '21

20 Jan '21

Hello, Kamailio 5.4.3 has been installed from source. "kamctl ping sip:10.10.10.10:5060" is sent from the command line. Errors: Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_mod.c:1352]: jsonrpc_exec_ex(): invalid json doc [[{"jsonrpc": "2.0", "method": "tm.t_uac", "], "reply_name": "kamailio_receiver_85791", "id": 85791}]].", "From:sip:daemon@KAM-TEST Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_fifo.c:461]: jsonrpc_fifo_server(): failed to execute the json document from fifo stream This command is working in 4.4. Has anyone tried using kamctl ping in 5.4? Is it a bug in 5.4.3 ? Thank you, Julia.

3 10

Header Manipulation with returned REST data
by Mark Lawrence 19 Jan '21

19 Jan '21

Hi All, Sorry for the intrusion and asking directly, but there’s so much information out there that I would spend years looking for the answer myself and spending equal amount of time in a lab playing. I do like the last idea but time is a little short. I’m thinking of using Kamailio to overcome a few downfalls in the main SBC/PBX commercial product space that I’ve typically deployed and wanted to ask if it is possible to use as a stateful proxy and carry out the following header manipulation type scenario with a bit more dynamassium: Use the FROM and TO header to create two value pairs that would be sent via a REST request to an external application, which will return two corresponding values to (a) supply a different value to update the original FROM header and (b) a full PAI sting to add on to the outgoing message. If you have experience of this or if it’s a nonstarter I’d love to know. There’s so many modules and great stuff that you have in your toy chest it is amazing and kudos to all you developers being so inventive out there: it leaves me to think I’ve been deploying SIP wrong all these years. Thanks and keep safe and well, Mark

2 1

Count the rows in a xavp from script
by Chaigneau, Nicolas 19 Jan '21

19 Jan '21

Hello, I'm looking for a way to retrieve the number of rows stored in a xavp. (Something better than manually iterating over the xavp) Basically, I'd like to call xavp_count from the script. Is that possible ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 1

Kamailio High Availability questions
by Andriano 17 Jan '21

17 Jan '21

Hello have some questions about Kamailio HA + loadbalancers (dispatcher modules) 1. is posible to make HA for RTP media? maybe RTPEngine +redis? is not possible to do this with RTProxy? where is some examples? 2. What do Kamailio db cluster module? it can replicate every DB query record like normal DB cluster? or better is to use native MySQL cluster mode? 3. How is working dispatcher modules in HA? how call is handled and htable in memory between loadbalancers endpoint nodes etc... IS it posible to do only with DMQ ? Just need understand how to do basic Kamailio setup with HA and load balancers in general, for example if will be 2 phisical servers, in every server is virtualisation and every server has 2 VPS. 1 VPS for frontend Kamailio loadbanancer, 1 VPS backend Kamailio app with accounts, routing logic, LCR, RTP media app etc... between load balancers i know maybe is possible to do floating ip, dns round robinetc... and if one VPs server down, call will be dropet in every situation? or call will not be droped and how to prevent it. Just need to know in general schema and architecture how to do this real HA in Kamailio and avoind sinlge point of failure and where to read and learn it ho to make HA in real. In the internet no many docs about it, many conference video how good it is, but no good examples with config, i did not founded, so will be very glad for any help in these newbie questions ;) with single kamailio is many answers, but is realy difficult to understand how to do voip HA with Kamailio without single point of failure. Thanks for any answers...

1 0

http_client module not found
by Joli Martinez 15 Jan '21

15 Jan '21

Hello, I am running kamailio 5.3.8 on Debian 9. Kamailio is installed via apt package manager. I am trying to get apiban.org to work and it seems to be having a problem with the http_client module. I have searched for the kamailio modules and I am unable to find anything that resembles http. I have also installed the extra module and no luck. What would be the correct module to install? kamailio: ERROR: <core> [core/sr_module.c:500]: load_module(): could not find module <http_client> in </usr/lib/x86_64-linux-gnu/kamailio/modules/>

2 2

Stubs for KEMI Javascript
by Nicki Otte 15 Jan '21

15 Jan '21

Hi everyone. I wanted all of you to know, that I have made JavaScript stubs, for autocompletion in IDE’s, for those of you that use KEMI JavaScript. Specifically tested on Jetbrains PHPStorm/Webstorm. The stubs are available for free at my GitHub page. The link is https://github.com/captainDuckay/Kamailio-javascript-stubs My main language isn't typescript or JavaScript, and this is one of my first public releases. With that said though, I am open for critics, or even guidance. I hope it's usable for you. // Nicki Bo Otte

2 3

To header being malformatted.
by AL S 14 Jan '21

14 Jan '21

Hi, We noticed an issue with the way the Kamailio is handling certain messages being sent to them, specifically, the To header being relayed from Kamailio to the next hop is malformatted. Any assistance is greatly appreciated INVITE received by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+12053140346@72.31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 INVITE forwarded by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+120531403Oh.lc".3/+0]1..31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 Another example in BYE: BYE received by Kamailio: To: "WIRELESS CALLER" <sip:+18131111111@71.44.13.84>;tag=gK0a788ca6 BYE forwarded by Kamailio: To: "WIRELESS CALLER" <sip:+18131112222A.++.+,-94 Thanks. AL

2 2

Source Port on TLS OPTIONS from Dispatcher
by Charles Phillips 14 Jan '21

14 Jan '21

Hello all. As they say in radio, “long time listener, first time caller” Anyway, I am having trouble getting past the following road block and any help would be greatly appreciated. Kamailio version is 5.4.3 When attempting to use dispatcher to send OPTIONS packets to several TLS destinations, the packets are leaving the Kamailio server on random ports. This is a problem because the servers I am sending the OPTIONS to (MS Teams) are enforcing rport so the responses are returned to a port that Kamailio is not listening on. I have tried to force the socket in the event route (relevant parts of snippet below) but it does not appear to help. I should also mention that I am not behind NAT and the TLS socket is specified in the dispatcher attrs. event_route[tm:local-request] { sip_trace(); $fs = “tls:**ip-address**:5061”; } I have used Kamailio as a TLS server for many projects, but this is my first time as a client. I am sure I am missing something. - Charles

4 10

BYE and TCP
by Kjeld Flarup 13 Jan '21

13 Jan '21

Hello I have a Kamailio running behind NAT, which sends calls to a VOIP service provider. I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router. Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog. To which port is the server supposed to send the BYE, and what field tells the server this. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

5 17

handling rejected REFER messages
by Jaime Gajardo 13 Jan '21

13 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux

2 1

handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

1 0

handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environmet. behind the nat we have 2 identycal nodes using REFER for transfer call to another destiny. in the other NAt side we have our public carrier. When some node send to kamailio the REFER message and kamailio reroute this message to the carrier. The call is rejected due company policies. We want know how can we do for convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

1 0

Re: [SR-Users] Using websockets with kamailio 5.4
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Daniel - Google is your friend. I use it often. https://github.com/havfo/WEBRTC-to-SIP

1 0

TSILO - transaction is not suspended
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Folks - I am attempting to set up a mobile push configuration, by following the published example(s). If we have more than one endpoint registered, this configuration works as expected (via the ts_store() path below). If we do not have another registered endpoint the transaction seems to be suspended but then cannot be resumed (t_suspend) after REGISTER arrives. INFO: <script>: [R-LOG]: INVITE from IP:[24.XXX.XXX.XXX:62665] [sip:jeremy2@stage][<null>] NFO: <script>: [R-SUSPEND]: Suspended transaction [41359:415220139] INFO: <script>: [R-SUSPEND]: fU:[jeremy] rU:[jeremy2] tU:[jeremy2] INFO: <script>: [R-SUSPEND]: fU:[jeremy] => tU:[jeremy2] INFO: <script>: [R-SUSPEND]: vtp:[41359:415220139] INFO: <script>: [R-SENDPUSH]: Push sent [(55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78][com][hq01gt5utegi18ipkh1b] [Node-Push]: About to push: 55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78 INFO: <script>: R-REGISTER]: jeremy2 registered INFO: <script>: [R-REGISTER]: Contact: [<sip:ilyartjr@qjwhdafxlbtk.invalid ;transport=ws>;expires=10;app-id=stage;pn-voip-tok=55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78] INFO: <script>: [R-REGISTER]: App id: stage INFO: <script>: [R-REGISTER]: Expires: 10 INFO: <script>: [R-JOIN]: jeremy2 INFO: <script>: [R-JOIN]: Resuming trasaction [41359:415220139] jeremy2 (41359:415220139) WARNING: tm [t_suspend.c:197]: t_continue_helper(): transaction is not suspended [41359:415220139] WARNING: tmx [tmx_mod.c:847]: w_t_continue(): resuming the processing of transaction [41359:415220139] failed Relvant kamailio.cfg [Note: I did notice we had to switch things up a bit if calls come from a PSTN number via SBC] modparam("htable", "htable", "vtp=>size=10;autoexpire=300") ... if (!lookup("location")) { send_reply("100", "Suspending"); route(SUSPEND); // *testing this path here * } else { route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-LOCATION]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); } ... route[SUSPEND] { if (!t_suspend()) { xlog("L_ERROR", "[R-SUSPEND]: Failed to suspend transaction [$T(id_index):$T(id_label)]\n"); send_reply("501", "Unknown Destinaton"); exit; } xlog("L_INFO", "[R-SUSPEND]: Suspended transaction [$T(id_index):$T(id_label)]\n"); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] rU:[$rU] tU:[$tU]\n"); if (isbflagset(FLB_PSTN)) { $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => rU:[$rU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$rU)]\n"); } else { $sht(vtp=>join::$tU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => tU:[$tU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$tU)]\n"); } } route[REGISTER] { if (!is_method("REGISTER")) { return; } if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); } if (!save("location")) { sl_reply_error(); } route(JOIN); } # Append branches or resume the transaction route[JOIN] { xlog("L_INFO","[R-JOIN]: $tU\n"); $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $var(hstored) = $sht(vtp=>stored::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if ($var(hjoin)==0) { if ($var(hstored)) { xlog("L_INFO","[R-JOIN]: Appending transaction ($tu)\n"); ts_append("location", "$tu"); exit; } xlog("L_INFO","[R-JOIN]: No INVITE to resume\n"); exit; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); xlog("L_INFO", "[R-JOIN]: Resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); exit; } # Lookup and relay after resuming transaction route[RESUME] { xlog("L_INFO", "[R-RESUME]: ru: [$ru] tu: [$tu]\n"); lookup("location"); route(SETUP_ROUTES); route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-RESUME]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); exit; } What did I miss?

4 6

Profile name as variable in set_dlg_profile
by Muhammad Sohaib 11 Jan '21

11 Jan '21

Is it possible to insert the current dialog into the profile by using the following way? modparam("dialog", "profiles_no_value", "mytest;xyz") $var(profilename)="xyz"; set_dlg_profile(" $var(profilename) "); Currently it is giving errors and not allowing to use variable instead of static profile name. Or is there any alternative to pass profile name as variable? --- Thanks,

4 6

Unload UAC remote registry
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm on Kamailio 4.4.7 and I'm trying to do uac.reg_refresh over a row non existing on DB to try to unload, but data still persist on Kamailio. How to unload only one registry from UACs remote registrations when delete a record from DB without do a uac.reg_reload? UAC remote registrations is above 2k rows, I don't know if it's much or not. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

3 2

Different responses for domain.dump
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there! I'm getting different responses from Kamailio 4.4.7 when domain.dump exec; From *kamcmd*: I'm getting two domains; .es and .net. Request to jsonrpc-s module: Response from jsonrpc-s module: And now, I'm getting only one domain. What's I'm doing wrong?

2 2

uac.reg_refresh command
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm playing with RPC on my Kamailio 4.4.7, and I getting rare behaivor with *uac.reg_refresh* command. When I run it thru kamcmd like docs describes, this is without *l_uuid* parameter, I'm geting "*error: 500 - Invalid Parameters*" output. And when I run with *l_uuid* parameter, I get no output. But in both cases, Kamalio don't refresh data from DB. But I run this command thru HTTP (JSONRPC-S), both call modes are equivalent. With or without l_uuid parameter geting no output and refresh data from DB. It's a normal behaivor? Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

2 3

list myself values
by Javier Valencia 11 Jan '21

11 Jan '21

Hi, a question from a Kamailio newbie. How can I list or show variable *myself* values? Regards, JV

2 2

UAC module registrations interval
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there, I'm using Kamailio 4.4.7 with UAC remote registrations module. In my database have all rows with expires column value's 3600, and my PBX (asterisk) min, max and default expires to 3600. But I can see how Kamailio it's sending REGISTER each 150segs. UAC module parameters values; modparam("uac", "reg_db_url", "mysql://XXX") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_contact_addr", "XXX") modparam("uac", "reg_timer_interval", 5) modparam("uac", "reg_retry_interval", 60) My uacreg table it's above 2k rows. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

2 1

how does rtjson work?
by Syed Shahryar 11 Jan '21

11 Jan '21

My use case is such that I want to be able to send a SIP MESSAGE destined for a single username to multiple users. In order to be able to map that target username to multiple usernames, I tried using rtjson and evapi/http (to generate the routing json), but none of them worked. Is it the right approach? Is there any other way to programmatically generate multiple targets for SIP request? I am familiar with dynamic routing but I need more flexibility. Initially, I tried sending the request to a different username but that didn't work and got the following message in the http async result route. kamailio | 19(25) ERROR: nathelper [nathelper.c:1461]: sdp_1918(): Unable to parse sdp body kamailio | 19(25) NOTICE: <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi msg branch kamailio | 28(34) NOTICE: {2 6661 MESSAGE q6hikd6tumsr1dh35kvj} <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi reply branch The message was received on the originally requested to user and also $tu still points to the original user instead of the new user even after executing the following code. KSR.rtjson.init_routes(response); KSR.rtjson.push_routes(); KSR.nathelper.fix_nated_contact(); In http response, I receive the following and don't see any rtjson errors when init_routes is executed on it. {"version":"1.0","routing":"parallel","routes":[{"headers":{"from":{"display":"test_1","uri":"sip: *test_1*@test.com:9908"},"to":{"display":"test_4","uri":"sip:*test_4*@ test.com:9908 "},"extra":""},"branch_flags":8,"fr_timer":5000,"fr_inv_timer":30000}]} Please note that the request was sent to *test_2*(a)test.com. I have tried other combinations of above but nothing works. I have already searched on Google and read relevant articles and blog posts. Thanks -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Lost MySQL connections - Error 500
by Duarte Rocha 11 Jan '21

11 Jan '21

Hi all, I've made a post last month regarding losing MySQL connections - https://lists.kamailio.org/pipermail/sr-users/2020-December/111389.html At the time I thought connections were dying as a consequence of low activity and traffic on the proxy. Meanwhile, I've migrated a great number of equipments to the proxy with Registers being refreshed every 10minutes and the problem still persists. In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. My SQL client on the Kamailio machine is mysql-community-client 5.6.50-2.el7. It writes and reads in a remote InnoDB database. This are the logs i get from Kamailio when the problem appears : Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lock wait timeout exceeded; try restarting transaction (1205) Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} <core> [db_query.c:348]: db_do_update(): error while submitting query Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1663]: update_contact_db(): failed to update database Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} registrar [save.c:784]: update_contacts(): failed to update contact Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Originally I had usrloc db_mode on mode 3 - DB-Only Scheme. In order to try to mitigate the issue I changed it to mode 1 - Write-Through scheme but even then I get the same log errors and an "500" error is still sent to the client. I chose this mode since, as far as I can understand it applies changes directly to DB but also uses cache. Please correct me if i'm wrong on that. Has this issue happened with anyone before? Is there a way to mitigate this issue? My only constraint is that I need the database to be always updated since I have an HA setup, and as such, I can't use cache only methods. Best Regards,

2 5

loose_route() - Local Proxy
by Duarte Rocha 08 Jan '21

08 Jan '21

Greetings, I'm using loose_route() from the RR module and i'm having troubles making it use the following exception from the code : "There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE." My example is a REGISTER without To-TAG which has a Route header with kamailio address. If i use proxy IP on Route header , loose_route() returns false as it should. However, if i use an hostname belonging to the proxy in the route, loose_route() returns true. I have hostnames and local ips defined in the "DOMAIN" table but it doesn't seem to be working. Which other places can local hostnames and ips be configured in order to be seen as local to loose_route() ? Best Regards

4 4

Contact header not removed properly Getting some junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my scenario call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org/> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060/>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.xx.xx.xxx 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.xx.xx.xxx 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 1

about Contact Header Junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my requirement call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.16.16.163 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.16.16.163 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 1

Dispatcher SNI Questions
by Tim Chubb 06 Jan '21

06 Jan '21

Hi All Is it possible to explicitly define the tls cert to be used per destination by dispatcher? I'm attempting to integrate with a service that requires domain name presented to match that of the cert. Where the use of dispatcher is probably not as intended anyway as it targets the same set of destinations multiple times (primary and secondary endpoints for the service im integrating with), with multiple distinct set id's (basically a set per customer for multi tenancy) which expects a sip ping per account to verify that connection is alive and healthy. #Example Dispatcher list #Account 1 1 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061 1 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061; #Account 2 2 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; 2 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; Currently I'm running a solution based around certs with multiple SAN (subject alternative name) defined, but this is a pain to administrate, and not as scalable as I would like. I want to be able to define multiple client:any tls profiles and explicitly send via that profile. Its easy enough to do that in config using the tls xavps to define a server name and/or id. But for the sip options ping by dispatcher to work you need to specify the server name or id before the tm:local-request event route fire's. I've resorted to hacking away at dispatcher to see what would happen if I set the tls server name or id before the OPTIONS message is sent, and it works great for the first tls client profile matched, but any others you define re-use the initial tcp connection so reuses the first connections tls config, thus presenting the wrong cert (I'm probably wrong but logs show only one tls complete_init() related to dispatcher pings and the second set's pings skips all the tls init logging so suggests some thing is cached/reused, and logging on the service indicates cert name mismatch with set1s cert being offered) So I've hit a bit of a dead end, is there a way to force a new tcp connection per server name/id I fear outbound SNI might be a bit of a can of worms, for it to work nicely with dispatcher as is, would be an extension of the proto:host:port format to include a name or id, maybe something like: tls(name=sub.domain.com):10.0.0.5:5061 But that looks like it would have a lot of wide reaching knock-on effects, so setting it as an attribute like ping_from works fine as I'm currently doing, if I could force a new connection from dispatch.c.... The other thing that's occurred to me is that it might be better to stop trying to adapt a small percentage of dispatcher's functionality and trying to hammer a round peg into a square hole and instead try and create a new module similar to dispatcher in so much that it sends pings (that's all I really need in this case) and use dispatcher as intended with a single set of non-repeated destinations for routing. This would also allow much simpler integration with MS teams (if added ability to set contact address on the background pings) as well as the service im trying to integrate, and a more natural data set to be used as in my example you can see the only difference is the customers domain between each set. How I would imagine it to work would be to create a tls connection per customer domain, then iterate through the connections per destination in pseudo json the data would look like this: { Destinations:[ [1,["a.cloudservice.com","b.cloudservice.com"]], [2,["sip.pstnhub.microsoft.com", "sip2.pstnhub.microsoft.com", "sip3.pstnhub.microsoft.com"] ], Sources:[ {uri:"sip.account1.com",destinationMapping:[1]}, {uri:"sip.account2.com",destinationMapping:[1,2]} ] } And would just run in the background sending pings with SNI support, and if based on dispatcher should make it easy enough adapt things like the destination up and down event routes, and would allow me to use Kamailio much like I would use nginx for web stuff as a reverse proxy and handle TLS offload on my edge gateway proxy. Any thoughts?, im frustratingly close to having something that will do if I could get a new connection to be created, but I fear my use case might be sufficiently different from dispatchers to make a new module a sensible approach.. Regards Tim.

2 1

Pre-loaded routes verification
by Duarte Rocha 04 Jan '21

04 Jan '21

Greetings, I'm doing a migration from a opensips system to kamailio one and i'm trying to replicate its functionalities. The system is a Registrar with a pstn gateway. I noticed that on the opensips version, loose_route() is called on initial requests and it seems to be used to protect against preloaded routes. However, on the default Kamailio configuration file loose_route() is only called on requests in-dialog and the verification mentioned above is not used. Why doesn't Kamailio includes this verification? Is there are any security concerns not using it? Best Regards,

2 1

using kamailio 5.4 with the websocket or xmlhttprequest
by Daniel Hermann N'don 04 Jan '21

04 Jan '21

Hello everyone, I'm a beginner in kamailio technology, and I would like to know how to communicate with kamailio from the web.

3 3

Unable to connect sip with Kamailio
by Rohit Kumar 04 Jan '21

04 Jan '21

Hello everyone, I'm beginner in Kamailio, I want to connect my asterisk Server with Kamailio. Can you please help me in this context. I shall be very thankful to you. Regards, Rohit Kumar On Thu, Dec 31, 2020, 16:31 <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 2. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Antony Stone) > 3. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 4. Using a specific log level for "script" ? (Chaigneau, Nicolas) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 30 Dec 2020 14:15:19 +0000 > From: Daniel Hermann N'don <danielhermann.ndon(a)outlook.com> > To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> > Subject: [SR-Users] using kamailio 5.4 with the websocket or > xmlhttprequest > Message-ID: > < > MWHPR05MB291167705F86195717CA7E3181D70(a)MWHPR05MB2911.namprd05.prod.outlook.com > > > > Content-Type: text/plain; charset="iso-8859-1" > > Hello everyone, I'm a beginner in kamailio technology, and I would like to > know how to communicate with kamailio from the web. >

2 1

Using a specific log level for "script" ?
by Chaigneau, Nicolas 04 Jan '21

04 Jan '21

Hello, Is it possible to set a specific log level for the Kamailio script ? I've seen that the < debugger > module allows to do that with modules or < core > : https://kamailio.org/docs/modules/5.4.x/modules/debugger.html#dbg.p.mod_lev… So I've tried to do : modparam("debugger", "mod_level_mode", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level", "my_module=3") // this works modparam("debugger", "mod_level", "core=3") // also works modparam("debugger", "mod_level", "script=3") // nope... But it doesn't seem to work for < script >... :/ Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

[Presence] publ_cache=2 usage
by Denys Pozniak 30 Dec '20

30 Dec '20

Hello! Could somebody explain how the parameters *publ_cache=2* and *subs_db_mode* depend on each other? Do I understand correctly that only the following modes of *subs_db_mode* are allowed 0,1,2? *subs_db_mode:0 - This disables database completely. Only memory will be used. Subscriptions will not survive restart. Use this value if you need a really fast presence module and subscription persistence is not necessary or is provided by other means.1 - Write-Through scheme. Subscriptions are updated synchronously in database and in memory(used for read operations). Use this scheme if speed is not top priority, but it's important that no subscriptions will be lost during crash or reboot or if you have an external application that reads the state of the subscriptions from database and they need to be updated synchronously.2 - Write-Back scheme. This is a combination of previous two schemes. All changes are made to memory and database synchronization is done in the timer. The timer deletes all expired contacts and flushes all modified or new subscriptions to database. Use this scheme if you encounter high-load peaks and want them to process as fast as possible. Latency of this mode is much lower than latency of mode 1, but slightly higher than latency of mode 0. To control the interval at which data is flushed to database, set the db_update_period parameter.3 - DB-Only scheme. No memory cache is kept, all operations being directly performed with the database. The timer deletes all expired subscriptions from database. The mode is useful if you configure more servers sharing the same DB without any replication at SIP level. The mode may be slower due the high number of DB operation.* -- BR, Denys Pozniak

1 0

Presence - issue with NOTIFY "terminated" state
by Zhan Bazarov 29 Dec '20

29 Dec '20

Hello everyone! I faced some unfortunately issue with presence module. Let me describe setup scheme a bit: I'm publishing dialog states via pua_dialoginfo from one kamailio server(for calls proccessing itself) to another one kamailio(standalone kamailio just for handling presence,blf,mwi etc) So, about issue, let's say we have device A which has outgoing subscription to extension which belongs to device B and device C(both devices has same extension but different usernames, webrtc+sip phone, effectively its the same Phone, I'm using pubruri_avps to change entities from usernames to extension) . And, in some cases,for examle, when thirtparty is making call to this extension(B,C devices) and making CANCEL after couple seconds - I see the PUBLISH from callserver to presence server with state "terminated', I see 200ok response.. but presence server isn't sending NOTIFY to device A about state "terminated", everything is good with "trying" and "early" states... That problem is happening 1 of 50 tries.. I've tried a lot of things to fix that - still are not succeed. p.s. everything is good from logs/sip point of view... And also, when I disable the webrtc device - I'm not able to reproduce that issue anymore. Looks like something wrong with case, when you have multiple different username behind the same presentity. Maybe someone had same issue and may help me. Thanks! -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Kamailio as a presence watcher
by Jukka Heikkilä 29 Dec '20

29 Dec '20

Hello, I'm working with IMS network and VoLTE related services. My problem is that I would like to use Kamailio as a presence watcher but I haven't managed to make it happen yet. The approach would be that once Kamailio gets 3rd party registration from S-CSCF the Kamailio would send SUBSCRIBE back to S-CSCF to subscribe the presence of subscription. Once a subscription place or get calls the status of subscription would be updated to Kamailio and to the database where it can be fetched. I have checked presence and pua modules, but haven't found a way to make it work as expected. Is there way to do that or is this totally incompatible approach to do that? Best Regards, Jukka Heikkilä jutsco(a)gmail.com

1 0

How to implement a self-sustained Presence server
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration? That is I want the Presence module to be fed status updates as Kamailio is routing calls and handling registrations. So when a user registers, his phone is ringing, he is on the phone, etc.., the Presence module gets notified and can update its subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

2 1

Kamailio suddenly stops
by Javier Valencia 28 Dec '20

28 Dec '20

Hi there! When I start my webrtc kamailio suddenly stops. I can see this on syslog: Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c2ab8), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c3138), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring ... omit vervose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c1f38), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_free(323) - ignoring ...omit verbose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557569f250), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_realloc(299) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557565cf60), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized OS info: root@webrtc:/etc/kamailio# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster root@webrtc:/etc/kamailio# free -m total used free shared buff/cache available Mem: 16041 134 15551 108 356 15524 Swap: 974 0 974 Kamailio version: version: kamailio 5.4.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 Kamailio params: SHM_MEMORY=512 PKG_MEMORY=32 SSL info: root@webrtc:/etc/kamailio# apt show openssl Package: openssl Version: 1.1.1d-0+deb10u4 Priority: optional Section: utils Maintainer: Debian OpenSSL Team <pkg-openssl-devel(a)lists.alioth.debian.org> Installed-Size: 1.495 kB Depends: libc6 (>= 2.15), libssl1.1 (>= 1.1.1) Suggests: ca-certificates Homepage: https://www.openssl.org/ Download-Size: 843 kB APT-Manual-Installed: no APT-Sources: http://security.debian.org/debian-security buster/updates/main amd64 Packages Description: juego de herramientas Secure Sockets Layer - utilidad criptográfica Este paquete forma parte de la implementación del proyecto OpenSSL de los protocolos criptográficos SSL y TLS para la comunicación segura a través de Internet. . Contiene el binario de propósito general para la línea de órdenes /usr/bin/openssl, útil para operaciones criptográficas como: * creación de parámetros clave RSA, DH y DSA; * creación de certificados X.509, CSR y CRL; * calcular resúmenes de mensajes; * cifrado y descifrado con cifras; * probar clientes y servidores SSL/TLS; * gestión de correo firmado o cifrado S/MIME. Should I use another version of openssl, or maybe use gnutls? Regards, Javier Valencia

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Self-sustained Presence server.
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration. That is I want the Presence module to be fed status updates from the same Kamailio that is routing the calls. So when a user registers, his phone is ringing, he is busy, on the phone, etc.., the Presence module gets notified and can update subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

1 0

Read $fd from mysql?
by Mihai Cezar 27 Dec '20

27 Dec '20

Hello, I am redirecting the agents/reps to their asterisk server based on the hostname which they connect, and, i wonder if it's possible to read from db instead of the file cause it's getting big ;) # Dispatch requests route[DISPATCH] { switch ($fd){ case "oro.streamlinepbx.nl": if (!ds_select_dst("1", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; case "oro3.streamlinepbx.nl": if (!ds_select_dst("2", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; ............ default: log("unknow destination?"); send_reply(503, "No service defined"); xlog("--- SCRIPT: going to <$ru> via <$du> ...Exiting"); exit; } xlog("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } Thanks,

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Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 27 Dec '20

27 Dec '20

A year that was very tough for the entire world is getting close to its end, no doubt that was the most challenging in our life so far for many of us, however, with a distributed, friendly and collaborative community, Kamailio project continued to develop consistently, bringing out a new major release. I trust it helped people and companies world wide to to connect with the dear ones and build a better way of living. The season holidays are ahead, time to rest a bit, spend time with family and friends, being online or face to face. As usual, I take this chance to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2020 another great year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

5 4

app_lua
by rob.van.den.bulk＠gmail.com 26 Dec '20

26 Dec '20

Hallo, I try to install the module app_lua in Kamailio 5.3. ls: cannot access '/usr/lib/liblua*': No such file or directory ls: cannot access '/usr/lib/liblua*': No such file or directory CC (gcc) [M app_lua.so] app_lua_mod.o In file included from app_lua_mod.c:32: app_lua_api.h:25:10: fatal error: lua.h: No such file or directory #include <lua.h> ^~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: app_lua_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/local/src/kamailio-5.3/kamailio/src' make: *** [Makefile:34: all] Error 2 Any help appreciated Greeting Rob van den Bulk

2 3

Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 23 Dec '20

23 Dec '20

FOSDEM - Real Time Communications devroom CfP ============================================= NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time slot. Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2021 takes place 6-7 February 2021 and for the very first time, it will be online. This document contains information about: - Real-Time Communications developer room (devroom) - speaking opportunities - volunteering New rules for the online edition -------------------------------- This year FOSDEM will be fully online instead of being held in Brussels, here are the most important things to know about this (quite significant) change: - The reference time will be Brussels local time (CET) - Talks will be pre-recorded in advance, and streamed during the event - The Q/A session will be live - A facility will be provided for people watching to chat between themselves - A facility will be provided for people watching to submit questions Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers who can help manage the scheduling and live Q&A sessions.** The devroom is only on Saturday, 6th of February 2021. To discuss the devroom, volunteer or ask questions, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Key dates - 20th Dec: Submission deadline (extended to 8 January) - 24th Dec: Announcement of selected talks - 15th Jan: Presentations upload deadline - 6th & 7th Feb: Conference dates (online) - 6th Feb: RTC devroom date (online) ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 20th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM21/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. This year this is more true than ever, being able to record your presentation offline without an audience in front can greatly help build up one's confidence! ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. ### Recording help The devroom organization is able to provide help with recording your session. The recording would be performed at a scheduled time with one of us, so you won't be alone giving your presentation. Minimal edits will be possible, but the ideal plan is to record it in one shot. Thanks Dan Jenkins for providing us with the means to do this! Volunteers needed ----------------- To make the devroom run successfully, we are looking for volunteers. This year many things be done for the first time, so all the help we can get is more than welcome. Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

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MID-Registrar Example
by moisesynfante 23 Dec '20

23 Dec '20

Hello, I'm a newbie when it comes to Kamailio and I would like to know if there's a working example of a Mid-Registrar implementation. I read on other posts that the UAC module can be used to achieve this but, honestly, I don't understand how. If any of you could provide some path I could follow I would be very thankful. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Re: [SR-Users] How to use port number using $env in listen
by Syed Shahryar 22 Dec '20

22 Dec '20

Any updates on this? On Tue, Dec 15, 2020 at 8:13 PM Syed Shahryar <syedshahryar(a)gmail.com> wrote: > I am new to Kamailio so may be doing something wrong. I looked at > https://www.kamailio.org/wiki/cookbooks/5.4.x/core. > > I tried the following combinations but getting errors in all cases: > > #!subst "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdef "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdefs "/EVAPI_PORT/$env(EVAPI_PORT)/" > > *Errors:* > 0(6) ERROR: <core> [core/pvapi.c:924]: pv_parse_spec2(): error searching > pvar "env" > 0(6) ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char > [E/69] in [$env(EVAPI_PORT)/] at [5 (5)] > 0(6) ERROR: <core> [core/re.c:170]: parse_repl(): bad specifier in > replace part /$env(EVAPI_PORT)/ > 0(6) ERROR: <core> [core/ppcfg.c:69]: pp_subst_add(): bad subst > expression: /EVAPI_PORT/$env(EVAPI_PORT)/ > > At a later step, I would like to do: > #!define EVAPI_LISTEN "0.0.0.0:EVAPI_PORT" > > And use EVAP_LISTEN in mod_param for the evapi module. > > And something similar to above for TLS settings. > > > On Tue, Dec 15, 2020 at 12:41 PM Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> as written on the tracker, look at using #!subst or #!substdef (see core >> cookbook), they should be able to get the substitution value from >> environment using $env(NAME). >> >> The master branch introduced #!defenv as a (simpler) alternative. >> >> Cheers, >> Daniel >> On 12.12.20 05:33, Syed Shahryar wrote: >> >> I tried many combinations of the following but I keep getting syntax >> errors: >> >> listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) >> >> >> TLS_IP and PUBLIC_IP are constants defined at the top using #!define. >> And, TLS_PORT is supposed to be the env variable. >> >> >> -- >> -- >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee, you should not >> disseminate, distribute or copy this email. Please notify the sender >> immediately by email if you have received this email by mistake and delete >> this email from your system. Email transmission cannot be guaranteed to be >> secure or error-free, as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender, >> therefore, does not accept liability for any errors or omissions in the >> contents of this message which arise as a result of email transmission. If >> verification is required, please request a hard-copy version. >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Funding: https://www.paypal.me/dcmierla >> >> > > -- > -- > This message contains confidential information and is intended only for > the individual named. If you are not the named addressee, you should not > disseminate, distribute or copy this email. Please notify the sender > immediately by email if you have received this email by mistake and delete > this email from your system. Email transmission cannot be guaranteed to be > secure or error-free, as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender, > therefore, does not accept liability for any errors or omissions in the > contents of this message which arise as a result of email transmission. If > verification is required, please request a hard-copy version. > -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 2

ndb_redis with keydb master/master
by David VILLAUME 22 Dec '20

22 Dec '20

Hello, I consider using keydb-server in an active/action setup without sentinel. Do you know if there is a way to declare both hosts in ndb_redis with the same server name (target use : topos) ? Regards, David

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Re: [SR-Users] DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 22 Dec '20

22 Dec '20

Apologies for not having included the sr-users in the previous email. Anyhow, the solution suggested by Social Boh indeed worked, as a side effect of which we would not run into the issue described. *NOTE:* There was a peculiar configuration in keepalived with keeping both the nodes as BACKUP and setting the 'nopreempt' flag only in the actual MASTER(one with higher 'priority') side to make the keepalived thing to work properly. However, I was wondering if someone could confirm if this issue has been fixed inside the source code itself as I am using an older release v5.3.2 Regards, Harneet Singh On Thu, Dec 17, 2020 at 9:52 PM harneet singh <hbilling(a)gmail.com> wrote: > Thanks Social Boh, > > I will investigate this possibility and if it does work in our case, that > should circumvent the issue. > In addition, is there a fix known to the SR-Users which could be available > inside the Dispatcher module to cause syncing just like the syncing is > there in the Dialog module via DMQ. I wanted to check this as I am using > v5.3.2 and unaware if new releases could have fixed this issue? > > Regards, > Harneet Singh > > On Wed, Dec 16, 2020 at 10:46 PM Social Boh <social(a)bohboh.info> wrote: > >> Maybe a option is leave the secondary like primary when primary go up >> newly. >> >> Example: Primary go down, Secondary is the new Primary. Primary came back >> up but Secondary still is the VIP and still process the calls. >> >> Only if Secondary go down, primary is the new VIP. >> >> The keepalived parameter for this behaviour is: >> >> *nopreempt* >> >> --- >> I'm SoCIaL, MayBe >> >> El 16/12/2020 a las 11:57 a. m., harneet singh escribió: >> >> Hi All, >> >> I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual >> IP) functionality, and have run into a rather peculiar issue. >> >> *Setup:* >> >> Caller ------------ KamailioVIP(*Primary *and *Secondary* >> Kamailio)------- Callee >> >> HA provided by Keepalived >> DMQ is used between the Primary and Secondary Kamailio for dialog sync >> >> *Issue Seen:* >> >> *Suppose the Primary Kamailio has been brought down* and the Secondary >> one is actually active and tied to the VIP. >> >> When a call is fired from the Caller, it traverses through the Secondary >> Kamailio and lands onto the callee. The dialogs are updated properly. At >> this point, the Primary Kamailio is brought up, and dialog state is synced >> due to the DMQ module. >> The Keepalived will now attach the VIP to the Primary Instance. If the >> caller hangs up the phone at this point, the BYE sip message traverses >> through the Primary Kamailio instance to the callee and the call is >> cleared, but there are two issues here. >> >> 1. The Primary Kamailio throws an error in *ds_load_remove() *saying >> that it cannot find the load for the specific call id. This is apparently >> due to the fact that the dispatcher data is not synced between the two >> modules but dialog data is. So dialog wise, things are fine. Can this be >> fixed somehow? >> >> 2. The above is still as grave a problem as the dialogs are cleared. >> BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* >> kamailio even well after the call is cleared, the *DLGLOAD* shows 1. >> Since we are using the *Call Load based distribution* for the >> dispatching, this is effectively one call stuck on the dispatcher, which >> leads to resource leak. >> >> Is this a known issue, and if so, do we have a way to circumvent this? >> >> Regards, >> Harneet Singh >> -- >> "Once you eliminate the impossible, whatever remains, no matter how >> improbable, must be the truth" - Sir Arthur Conan Doyle >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> > > -- > "Once you eliminate the impossible, whatever remains, no matter how > improbable, must be the truth" - Sir Arthur Conan Doyle > -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

1 0

Kamailio v5.3.8 released
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, Kamailio SIP Server v5.3.8 stable has been released. This is a maintenance release of the previous stable branch (5.3), that includes fixes since the release of v5.3.7. There are no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are recommended to be upgraded to v5.3.8. For more details about version 5.3.8 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/12/kamailio-v5-3-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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MSTeams Kamailio with PBX Hold Refer issue
by me.projects＠yahoo.com 21 Dec '20

21 Dec '20

Hi, I have been trying to integrate MSTeams with PBX Through Kamailio. Calls are working now. Hold has Issues. MSTeams uses REFER to manage this all with NOTIFY. I am sending t_send_reply("202", "Accepted"); when receive Refer now I need to send NOTIFY I think I can use t_uac_send() NOTIFY I am unable to send t_uac_send properly on when recertifications refer from MSTeams. Any one with the experience on this ? Thanks,

2 1

Trying to use $sel(contact.uri.params)
by Chaigneau, Nicolas 21 Dec '20

21 Dec '20

Hello, I'd like to use a < select > as described here: http://www.kamailio.org/wiki/cookbooks/5.4.x/selects#contactparams_s I'm trying to extract a parameter from the Contact URI. What I've tried to do: xlog("L_INFO", "contact.uri: $sel(contact.uri)\n"); # OK xlog("L_INFO", "contact.uri.params: $sel(contact.uri.params)\n"); # null ?? This does not seem to work... see output here : 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri: sip:+33145678901;tgrp=YC031;trunk-context=YC90@10.31.22.2:5099 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri.params: <null> What am I doing wrong ? Maybe this is the wrong approach for what I'm trying to do ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

Releasing Kamailio 5.3.8
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, I am planning to release next week (likely on Monday 21st December) the next minor version out of branch 5.3, version 5.3.8. Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker). Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

send MESSAGE to other sessions of same user
by Syed Shahryar 21 Dec '20

21 Dec '20

What's the best way of sending a message to all other sessions (and the receiver) of the same sender? For example: If user1(a)example.com is logged in to Kamailio from multiple devices and from one of the devices user1 sends a SIP message to user2(a)example.com. I want the message forwarded not only to user2(a)example.com but also to all other user1(a)example.com sessions. What's the easiest way to achieve this? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

1 0

Kamailio not updating CSeq after uac_auth() for ACK
by Terry Tailor 18 Dec '20

18 Dec '20

Hi, Here is setup details Kamailio Proxy a <-----> Kamailio Proxy b <-----> Asterisk Kamailio proxy b auth based on IP Kamailio Proxy a 1. Kamailio proxy sends an Invite request to Kamailio proxy b with CSeq: 102 INVITE 2. Kamailio proxy b check again IP auth and sends invite request to asterisk with CSeq: 102 INVITE 4 asterisk response with 401 Unauthorized with CSeq: 102 5. Kamailio proxy b sends ACK request with CSeq: 102 ACK 6. Kamailio proxy b sends INVITE request with CSeq: 103 INVITE 7. Asterisk sends 100 trying and 180 ringing with CSeq: 103 INVITE 8. Kamailio proxy b relay 180 ringing with CSeq: 102 INVITE 9. Asterisk send 200 Ok with CSeq: 103 INVITE and proxy b relay it to proxy a with CSeq: 102 INVITE 10. Now Proxy a send an ACK with CSeq: 102 ACK and proxy b relay ACK to asterisk with CSeq: 102 ACK Now asterisk is retransmitting 200 ok with CSeq: 103 INVITE and proxy a is sending ACK with Cseq: 102 ACK Same happing in case of 487 Request Terminated Here is my config modparam("dialog", "default_timeout", 10800) # 3h modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "send_bye", 1) modparam("dialog", "track_cseq_updates", 1) modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 3 sec modparam("tm", "fr_timer", 2000) modparam("tm", "retr_timer1", 250) modparam("tm", "retr_timer2", 500) # default invite retransmission timeout after 1xx: 60 sec modparam("tm", "fr_inv_timer", 60000) modparam("tm", "reparse_on_dns_failover", 0) if (is_method("INVITE") && !has_totag()) { dlg_setflag(FLD_RTPPROXY); dlg_manage(); } route(RELAY); route[RELAY] { # At this point all checks are passed t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); t_on_failure("MANAGE_FAILURE"); if(is_method("INVITE") && isflagset(FLT_FROMDID)) { #$fd = "sip.comms.kuflink.com"; t_on_failure("RELAY_AUTH"); } if (!t_relay()) { send_reply("500", "Cannot relay to $du/$ru"); } exit; } failure_route[RELAY_AUTH] { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { #$fd = "sip.comms.kuflink.com"; $avp(auser) = "DID_UA_USERNAME"; $avp(apass) = "DID_UA_PASSWORD"; if (uac_auth()) { t_relay(); } exit; } } route[WITHINDLG] { if (!has_totag()) { return; } if (is_method("BYE")) { rtpengine_delete(); } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessly route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK") ) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } send_reply("404","Not here"); exit; } kamailio -v version: kamailio 5.3.7 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 uname -a Linux example.com 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 GNU/Linux SIP logs 2020/12/17 10:08:18.483907 proxy-a.com:5060 -> 10.0.14.118:5060 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.485063 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1;rport=5060 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 2020/12/17 10:08:18.485576 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.622867 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip1.comms.kuflink.com", nonce="11139d33" Content-Length: 0 2020/12/17 10:08:18.624034 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@proxy-a.com SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK Content-Length: 0 2020/12/17 10:08:18.625557 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 103 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 Authorization: Digest username="xxxxx", realm="xxxx", nonce="11139d33", uri="sip:123456@proxy-a.com", response="qwewqeqwe", algorithm=MD5 2020/12/17 10:08:18.779483 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.937756 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.938593 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Length: 0 2020/12/17 10:08:22.075814 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.077049 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.103365 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.103877 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.79f351d031912aae585207d2f26d5650.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:22.575533 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.576777 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.581649 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.582313 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.33626b5695ad4648ba393bc62bed6a8c.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:23.575828 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:23.577863 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316

4 7

Kamailio RTP proxy require?
by Mihai Cezar 17 Dec '20

17 Dec '20

Hi, Can't figure it out when to use RTPproxy, i am forwarding everything to asterisk in seem to work without RTPproxy. The scenario is like this: Softphone -> INTERNET -> Kamailio - > Asterisk (private ip) Now it seems to work without any rtpproxy but is it ok?

5 10

DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 16 Dec '20

16 Dec '20

Hi All, I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual IP) functionality, and have run into a rather peculiar issue. *Setup:* Caller ------------ KamailioVIP(*Primary *and *Secondary* Kamailio)------- Callee HA provided by Keepalived DMQ is used between the Primary and Secondary Kamailio for dialog sync *Issue Seen:* *Suppose the Primary Kamailio has been brought down* and the Secondary one is actually active and tied to the VIP. When a call is fired from the Caller, it traverses through the Secondary Kamailio and lands onto the callee. The dialogs are updated properly. At this point, the Primary Kamailio is brought up, and dialog state is synced due to the DMQ module. The Keepalived will now attach the VIP to the Primary Instance. If the caller hangs up the phone at this point, the BYE sip message traverses through the Primary Kamailio instance to the callee and the call is cleared, but there are two issues here. 1. The Primary Kamailio throws an error in *ds_load_remove() *saying that it cannot find the load for the specific call id. This is apparently due to the fact that the dispatcher data is not synced between the two modules but dialog data is. So dialog wise, things are fine. Can this be fixed somehow? 2. The above is still as grave a problem as the dialogs are cleared. BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* kamailio even well after the call is cleared, the *DLGLOAD* shows 1. Since we are using the *Call Load based distribution* for the dispatching, this is effectively one call stuck on the dispatcher, which leads to resource leak. Is this a known issue, and if so, do we have a way to circumvent this? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 1

INFO: RTC Devroom Fosdem 2021 - Online Event
by Daniel-Constantin Mierla 16 Dec '20

16 Dec '20

Hello, (cross-posting to kamailio public groups, a few may get this message duplicated). Fosdem 2021 is going to be online, meaning that it opens the possibility for everyone to do a presentation on the RTC Devroom, no longer being required to travel to Brussels. Therefore I am sending here the link to CFP in case someone would be interested to present about Kamailio or something else RTC-related: * https://github.com/FreeRTC/fosdem/blob/master/2021/cfp.md New speakers are always welcome! Only have in mind that the event is targeting developers and source code development, therefore your proposals have to be made from this perspective. Note also that submission deadline is 20th of December, very short time frame from now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

ERROR: dmq [notification_peer.c:337]: add_server_and_notify(): error adding notification node
by Syed Shahryar 16 Dec '20

16 Dec '20

Here is my setting: modparam("dmq", "server_address", "sip:0.0.0.0:5090" ) modparam("dmq", "notification_address", "sip:0.0.0.0:5090") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 2) modparam("dmq", "worker_usleep", 0) modparam("dmq", "ping_interval", 90) I have tried localhost and I believe other options but to no avail. I also get the following message: ERROR: dmq [dmq_funcs.c:516]: ping_servers(): cannot retrieve initial nodelist from sip:0.0.0.0:5090 How do I set up DMQ for adding and removing any Kamailio instance to/from the cluster and any Kamailio instance could serve any request coming from any client? ie. all Kamailio instances keep the current state in DMQ. I also have dmq_usrloc enabled. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Kamailio dtmf signaling
by Amit Pal 16 Dec '20

16 Dec '20

Dear Team, I am using kamailio + rtpengine, how can I processed dtmf from cfg scripts. even thought I can able to see dtmf over syslog. but can not able to capture in kamailio.cfg. how can I prepared ivr using kamalio? Please find the attached syslog. Regards Amit Pal

3 2

ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri </sip?jwt=..
by Syed Shahryar 15 Dec '20

15 Dec '20

I am sending JWT tokens via SIP URL through JSSIP for authentication before websocket connection is made. However, I am getting errors like: ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri < */sip?jwt=..* I have the following setting in kamailio config: modparam("xhttp", "url_match", "^/sip") // only match this path Is this a bug or some configuration issue? How do I resolve it? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

X-Forwarded-for IP to asterisk
by Mihai Cezar 15 Dec '20

15 Dec '20

Hello, Is it something like the XFF for Kamailio so I can see in asterisk (sip show peers) the ip of the registered softphone and not Kamailio's IP ? Thanks!

4 5

How to use port number using $env in listen
by Syed Shahryar 15 Dec '20

15 Dec '20

I tried many combinations of the following but I keep getting syntax errors: listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) TLS_IP and PUBLIC_IP are constants defined at the top using #!define. And, TLS_PORT is supposed to be the env variable. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Losing MySQL connections
by Duarte Rocha 15 Dec '20

15 Dec '20

Greetings all, I'm having some dropped MySQL connections in my Kamailio in times of low or no activity in the proxy. These are the errors i'm getting : db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) <core> [db_query.c:348]: db_do_update(): error while submitting query usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed usrloc [ucontact.c:1663]: update_contact_db(): failed to update database registrar [save.c:784]: update_contacts(): failed to update contact sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. Is this the best approach? I've restarted my Kamailio but i don't see any ping going to my BD. Is there any other config that I need to add so that the pings are sent? Is it possible to do a ping to verify connection and reconnect mannualy on the config file? Best Regards,

2 1

Problems debugging memory issues
by Sheldon L 15 Dec '20

15 Dec '20

Hi, I'm trying to investigate a possible memory leak issue in Kamailio and I'm following the instructions indicated in https://www.kamailio.org/wiki/tutorials/troubleshooting/memory I'm using Kamailo 4.4 with the following kamailio internals: Print out of kamailio internals Version: kamailio 4.4.2 (x86_64/linux) 892ad6 Default config: /apollo/env/TachyonPlatformKamailio/etc/kamailio/kamailio.cfg Default paths to modules: /apollo/env/TachyonPlatformKamailio/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_LISTEN=16 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 892ad6 Compiled with: x86_64-unknown-linux-gnu-gcc 4.9.4 Compiled on: 23:36:48 Dec 8 2020 Thank you for flying kamailio! I have attached to one of the kamailio tcp receiver processes via GDB and I am trying to dump memory blocks from the memory manager. However, mem_block is not in the current_context. The following is the output from my gdb session: Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-amazon-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /opt/codedeploy-agent/deployment-root/95bb1634-e653-4cec-b327-99b18b9227a8/d-7AJ1XZP58/deployment-archive/sbin/kamailio...done. Attaching to program: /apollo/env/TachyonPlatformKamailio/sbin/kamailio, process 8946 ptrace: Operation not permitted. (gdb) set $i=0 (gdb) set $a = mem_block->first_frag No symbol "mem_block" in current context. (gdb) info list Undefined info command: "list". Try "help info". (gdb) info threads No threads. Is there something I need to do before accessing the mem_block symbol? Also do you have any other suggestions for debugging memory leaks that may not be documented? Note: i can't upgrade to the latest version of Kamailio as we have custom modules and would require considerable effort to upgrade. Thanks, Sheldon.

2 1

sngrep TLS connection
by Agiftel 15 Dec '20

15 Dec '20

Hi all, does anyone was able to use sngrep (with -k option) to decrypt TLS connections? I'am doing what help says; used private key is the correct one; but no traffic is seen on sngrep. If i use not ecrypted call i can see everything, so sngrep is working good. Any hint? regards -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

5 8

Kamilio+asterisk = deregister after 2-3 minute
by Mihai 15 Dec '20

15 Dec '20

Hi, I am forwarding all the traffic to my asterisk server ( including registers) everything seems to work when i first register but if stay idle 2-3 minute it gets unregistered and then i can't register back unless i restart kamailio see attached print screen

6 6

voltdb support
by Ali Taher 14 Dec '20

14 Dec '20

Hello, I just need to know if kamailio can communicate with Voltdb for sqlops operations for example. Regards, Ali Taher

2 8

Kamailio v5.4.3 Released
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, Kamailio SIP Server v5.4.3 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.3. For more details about version 5.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/12/kamailio-v5-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Wrong To tag in t_reply
by Ivan Ribakov 14 Dec '20

14 Dec '20

Hi all, I’m trying to implement following routing behaviour: - ingress INVITE is sequentially forked to 1 or more endpoints - under certain conditions sequential forking may be stopped and pre-defined error returned to the caller Here’s is part of the config responsible for handling errors: failure_route[ROUTE_ELEMENT_FINAL_FAILURE_HELPER] { route(ROUTE_ELEMENT_FINAL_FAILURE); } route[ROUTE_ELEMENT_FINAL_FAILURE] { # We need to check if the error happens before the first egress, because T-state is not defined yet so we can not use t_reply if ($var(before_first_egress) == 0) { t_reply("$var(sipCode)", "$var(sipMessage)"); } else { sl_send_reply("$var(sipCode)", "$var(sipMessage)"); } } Here is a simple scenario where delivering call to first endpoint times out after receiving 180 Ringing (using custom fr_inv_timer value) where: - .33 is caller - .208 is Kamailio - .138 is callee Notice that 504 is retransmitted by Kamailio. Here is a comparison of 180 and 504 returned by Kamailio: As can be seen, 180 was proxied from the callee and has callee’s to-tag (tag=CALLEE1), however 504 that was generated by Kamailio using t_reply() has a different to-tag (presumably generated by Kamailio). How can I tell Kamailio not to generate new to-tag but to use the last one that proxied back to the caller? Thanks in advance for any help! Regards, Ivan

2 2

Releasing Kamailio v5.4.3
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, I am considering to release Kamailio v5.4.3 sometime next week, likely on Monday, December 14, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

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