sr-users May 2021

sr-users@lists.kamailio.org

57 participants
66 discussions

No Audio For Outbound Calls
by Kashish Raheja 28 May '21

28 May '21

Hi All, I have set up Kamailio in the following manner: Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) ---> Asterisk Server (on Cloud having public IP) I am successfully able to route the call to Asterisk server on Cloud when I make a call to the number provided by the carrier and there is audio also on both sides. However, when I am making an outbound call from Asterisk server to the number through Kamailio, there is no audio when I pick up the call. I have tried to capture the traces but not able to understand the exact problem here. Note: I am running the RTP proxy on Kamailio server. Any help on why this might be happening? Thanks. Regards Kashish +919413745250

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kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 28 May '21

28 May '21

Hello, I have question regarding of call forking on kamailio. Maybe somebody can help me. For example I have table like +--------------+---------------+--------------+---------------+--------------+ | First number | Second number | Third number | Fourth number | Fifth number | +--------------+---------------+--------------+---------------+--------------+ | A number | B number | C number | --- | --- | +--------------+---------------+--------------+---------------+--------------+ Table 1: Extra_numbers or separated by comma +--------------------+-----------------------------+ | First_number | Additional_numbers | +--------------------+-----------------------------+ | Primary number | 2nd,3rd number | +--------------+-----------------------------------+ Table 2: Extra_numbers +--------------------+---------------------------------+ | First_number | Additional_numbers | +--------------------+---------------------------------+ | Primary number | 2nd number,3rd number,4 number | +--------------------+---------------------------------+ Table 3: Extra_numbers If kamailio get the call (INVITE) to some number/extension (for example, "1st number"), then in routing I request (select `Additional_numbers` from `Extra_numbers` where First_number='1 number';) from the mysql database and there I check, if "Additional_numbers" exists in the database. Then I add "Additional_numbers" to the routing, and generate INVITE to "Additional_numbers" and 1 number. (always) set r-uri ($ru) to first destination (2 number) set r-uri ($ru) to second destination (3 number) set r-uri ($ru) to Third destination (1 number) set r-uri ($ru) to n destination How could this be done? Who can help with routing? -- Best regards, Fjodor Izmajlov

3 2

Setting the prefix in Json formatted logs
by Abdirahman Osman 27 May '21

27 May '21

Hello everyone, Tried to use the log-engine to push json formatted logs, here is a what kamailio is sending, my question is, How can I set the kamailio prefix? (232) { "kamailio.level": "DEBUG", "kamailio.module": "tm", "kamailio.file": "timer.c", "kamailio.line": 634, "kamailio.function": "wait_handler", "kamailio.callid": "", "kamailio.logprefix": "", "message": "finished transaction: 0x7f232236b1c0 (p:0x7f232216ae58/n:0x7f232216ae58)" } Thank you! Abdirahman

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kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 27 May '21

27 May '21

1 0

Body formats for filter_body()
by Duarte Rocha 27 May '21

27 May '21

Hello all, A couple years ago the function filter_body() from textops would only work with "application/sdp" if the string on the body was "Content-Type: application/sdp\r\n". It would fail if there wasn't a space after ":" and it would fail if there was present "\r\n\r\n" instead of "\r\n". Is this the actual behaviour of the function or has it been changed to also identify SDP in the formats mentioned above? Best Regards,

2 1

Collecting stats from Kamailio.cfg using Kamailio_exporter Prometheus
by Hisham Ahmad Jan 26 May '21

26 May '21

Hi, I'm using Kamailio version 5.3.7 with DISPATCHER module and I would like to integrate my kamailio with a kamailio_exporter for monitoring using Prometheus server. Is it possible to collect data such as the number of total calls (inbound, outbound) and plot time-graphs using those stats? If so, what modules/functions would I need apart from the Dispatcher module to be able to have efficient results? Any degree of help will be appreciated. *Hisham Jan* *VoIP Engineer* *0322-6398775*

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not able to display number in SCA module
by Aatif Shaikh 26 May '21

26 May '21

Hello All, I am using sca module to handle SLA presence and I am using sca.update_appearance RPC command to update the display name. kamcmd sca.update_appearance sip:215@voice.example.com 3 held KELVIN after executing the above command I am getting the KELVIN name on my Polycom phone display but I want to also update the display number so is there any way to update the display number for SLA presence using the SCA module

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segfault kamailio 4.4.6 Registrar
by Jonathan Hunter 26 May '21

26 May '21

Hi Guys, We are running kamailio 4.4.6 on debian 8 and not had any issues, however we did suffer a segfault and a core was generated; May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: ALERT: <core> [main.c:740]: handle_sigs(): child process 9575 exited by a signal 11 May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: ALERT: <core> [main.c:743]: handle_sigs(): core was generated May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD Running gdb on the core file generated returns this output; Reading symbols from /usr/sbin/kamailio...Reading symbols from /usr/lib/debug/.build-id/bc/fbd7e44c057e9e7e0680c2a5e8f6a47290c5ba.debug...done. done. [New LWP 9575] warning: Error reading shared library list entry at 0x28250c3348 warning: Error reading shared library list entry at 0x79c0850018bf73e8 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x0000000000589cce in lval_pvar_assign (lv=<optimized out>, rv=<optimized out>, msg=<optimized out>, h=<optimized out>) at lvalue.c:353 353 lvalue.c: No such file or directory. (gdb) bt full #0 0x0000000000589cce in lval_pvar_assign (lv=<optimized out>, rv=<optimized out>, msg=<optimized out>, h=<optimized out>) at lvalue.c:353 r_avp = 0x6 destroy_pval = -874501672 pvar = 0x0 pval = {rs = {s = 0x0, len = 0}, ri = 0, flags = 0} avp_val = {n = -874501704, s = {s = 0x7fb6cbe029b8 "\272/}", len = 5807505}, re = 0x7fb6cbe029b8} v = 6 #1 lval_assign (h=0x7fb6cbe029d8, msg=0x344, lv=0x0, rve=0x0) at lvalue.c:400 rv = 0x7fb6cbe029b8 ret = -874501672 __FUNCTION__ = "lval_assign" #2 0x0000000000000000 in ?? () No symbol table info available. Is this of any help? Could someone advise what next steps might be? Should we be looking to upgrade or is there a way to get more detail out of the core dump? Many Thanks Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

3 2

Kamailio 5.3.8 and RTPengine 9.4.1.1+0~mr9.4.1.1
by Mihai Cezar 23 May '21

23 May '21

Hi All, I've been using Kamailio as a proxy for asterisk (with public ips), everything was fine until i've changed the setup to private ips for asterisk. What I did so far: - Kamailio and rtpengine are on a box with 1 public IP. - Asterisk containers on private ips each asterisk has 2 trunks ( 1trunk voip provider 1trunk the Proxy). What works: - Proxy registrations and sip signaling works. What it does not work: - On the asterisk I got only on rtp connexion, my guess is that has something to do with the SDP, asterisk put the private ip in the SDP header. RTPengine is configured using this guide https://github.com/sipwise/rtpengine Can you have a look at the kamailio config, i think i am missing something or do something wrong. Thank you! command to start was: rtpengine --table=42 --listen-ng=127.0.0.1:2223 --interface=public/110.20.20.1 --tos=184 --pidfile=/run/rtpengine.pid --no-fallback and the Kamailio cfg is this: #!KAMAILIO #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define WITH_DEBUG #!define WITH_NAT ## This is the main configuration file for our proxy! ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 memdbg=5 memlog=5 log_stderror=yes disable_core_dump=no sip_warning=yes #!else debug=-0 log_stderror=no sip_warning=no disable_core_dump=yes #!endif log_facility=LOG_LOCAL0 fork=yes children=8 auto_aliases=no port=5060 disable_tcp=yes server_id = 1 server_signature=yes server_header="Server: proxy" dns_try_ipv6=no dns_try_naptr=no dns_retr_time=1 dns_retr_no=1 rev_dns=no ####### Modules Section ######## #loadmodule "db_mysql.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 "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "path.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" ####### Modules Settings # ----------------- setting module-specific parameters --------------- modparam("xlog", "buf_size", 4096) modparam("xlog", "prefix", "xlog: ") modparam("xlog", "log_facility", "LOG_DAEMON") modparam("xlog", "log_colors", "L_ERR=cr") modparam("xlog", "log_colors", "L_ERR=cr;L_WARN=px") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) #!ifdef WITH_NAT modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@proxy") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "rtpengine_retr", 2) modparam("rtpengine", "hash_table_size", 256) modparam("rtpengine", "hash_table_tout", 600) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "max_expires", 3600) modparam("registrar", "gruu_enabled", 1) # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/plugins/dispatcher.list") modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:ping@proxy") modparam("dispatcher", "ds_probing_mode", 1) modparam("path", "use_received", 1) ####### Routing Logic ########### # main request routing logic request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); if(ds_is_from_list()) { setflag(FLT_FS); } # CANCEL processing if (is_method("CANCEL")) { rtpengine_delete(); 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); if (isflagset(FLT_FS)) { route(FROM_WORLD); exit; } ### only initial requests (no To tag) # 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 sl_send_reply("100","Trying"); } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); } route[FROM_WORLD] { record_route(); route(RELAY); exit; } route[RELAY] { if (!t_relay()) { sl_reply_error(); } if (!add_path()) { sl_send_reply("503", "Internal Path Error"); } 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 (has_body("application/sdp")) { rtpengine_manage(); } append_hf("Supported: path\r\n"); exit; } # Per SIP request initial checks route[REQINIT] { 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("1511", "7")) { xlog("L_WARN","Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { rtpengine_delete(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); # exit; } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); } if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); }; route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not found"); exit; } # Dispatch requests route[DISPATCH] { # to add more servers in k8s if (!ds_select_dst("1", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("3")) { if (is_method("REGISTER")) { xlog("L_WARN", "natdetect fix_nated_register ip: $si\n"); fix_nated_register(); } else { xlog("L_WARN", "natdetect set_contact_alias ip: $si\n"); fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { xlog("L_WARN", "sdp fix ip: $si\n"); fix_nated_sdp("10"); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { xlog("L_WARN", "nat_uac_test 1..."); rtpengine_manage("replace-origin replace-session-connection direction=public ICE=force"); } else { xlog("L_WARN", "nat_uac_test addrtp..."); rtpengine_manage("codec-strip-all codec-transcode-PCMA codec-transcode-PCMU codec-transcode-G729 replace-origin replace-session-connection direction=public ICE=force "); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_WARN", "new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_WARN", "incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { rtpengine_manage(); } } onreply_route[REPLY_ROUTE] { if(status=~"2[0-9][0-9]") { rtpengine_answer("replace-session-connection replace-origin"); } if (has_body("application/sdp")) { rtpengine_offer(); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_WARN", "failure route\n"); route(NATMANAGE); if (t_is_canceled()) exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xlog("Retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

3 3

Can Kamailio help with this?
by Antony Stone 22 May '21

22 May '21

Hi. I've been aware of Kamailio, and on this list, for several years, but so far I have not implemented Kamailio for any purpose. I wonder if it could be a suitable tool for the following scenario. I currently use Asterisk to register as a client, with username and password, to other SIP PBXs in order to receive calls. As far as the other PBX is concerned, Asterisk looks like a SIP telephone on a particular extension. However, Asterisk's SIP client capabilities are limited to handling phone calls only, and it specifically cannot receive presence information about the state of other extensions from the remote PBX. So, could Kamailio be used in the path between Asterisk and the remote PBX so that the registration on the PBX comes from Kamailio (with any incoming calls being passed to Asterisk), but with Kamailio also receiving presence information from the remote PBX and making this available to some script or application (which I would expect to have to write)? The main point is that I don't want to have two things registering to the remote PBX, one for calls and one for presence, so I'm looking for a way to register with something which understands presence, and can also pass calls on to Asterisk. Can anyone suggest whether Kamailio could do this, and if so, point me at some resources to help me get started; or alternatively suggest ideas on some other tool which might be more appropriate than Kamailio? Thanks for any ideas, Antony. -- Schrödinger's rule of data integrity: the condition of any backup is unknown until a restore is attempted. Please reply to the list; please *don't* CC me.

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sr-users May 2021