- sr-users - kamailio.org

Using sngrep for visualising encrypted SIP traffic
by Ovidiu Sas 24 Mar '21

24 Mar '21

Hello all, Here's a quick recipe to ease the troubleshooting of encrypted SIP traffic: https://voipembedded.wordpress.com/2021/03/22/troubleshooting-kamailio-encr… Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

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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

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