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9 participants
37392 discussions

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?

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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

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