- sr-users - kamailio.org

multihomed kamailio and sip-reinvites SRTP to RTP
by Karsten Horsmann 07 Nov '19

07 Nov '19

Hello Mailinglist, i try to figure out, how to solve RE-INVITES and SRTP: i have an kamailio route names route[MEDIAPROXY] that i used to controll rtpengine in an private and public ip setup. This works fine for me now. It use the permission module and the adress table to figure out $var(mediagroup) = allow_address_group($Ri, 0); received from internal or external ip. The idea is from SBC-OS chamless stolen, so thanks for that. So fare so good. I adapted SRTP also to this. But on RE-INVITES it stops handling SRTP correctly to the encryption upstream. And i am sure, its my fault. But i have no clue how to detect that correctly. Any hints where i can read more about an correct handling on this? Here my main-config file / kamailio 5.3.x and rtpengine. https://gist.github.com/khorsmann/02bd98a628a190caee68fc3d876a3448 -- Mit freundlichen Grüßen *Karsten Horsmann*

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Re: [SR-Users] Group module
by Daniel Tryba 07 Nov '19

07 Nov '19

On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote: > > Hm... Maybe there is other module to achieve such functions? > Best if it would be with cache :) > But if no - regex also fine. But if it works it works! I don't think there is a specific module to do this but it is easy to implement yourself. What I don't like personally about your solution is the use of the permission tables, you need atleast 1 per user (unless users have the same ip). Personally I do this by direct database queries in usr_preferences on INVITEs and REGISTERs. You could use any database (depending on how you store the ACL) and cache it in a htable. My solution is mysql specific (inet_aton) and forces users to have atleast 1 CIDR notated subnet: if(!avp_db_query("select value from usr_preferences where username='$au' and attribute='acl' and inet_aton(substring_index(value,'/',1))&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)=inet_aton('$si')&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)")) { t_reply("404","Go away"); exit; }

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MongoDB findone(). Is projection supported?
by Lars Olsson 06 Nov '19

06 Nov '19

Hi, I am using NDB_MONGODB to ask some questions to a database. My database structure is pretty heavy, so I like to return as little data as required to Kamailio. I have tried the following MongoDB syntax for projection to include only given (my_field) column in the response. mongodb_find_one("srv_name", "database", "collection", "{ \"user_id\" : \"$var(user_id)\" }, { \"my_field\" : 1}", "mgr1") I always get full DB entry back. Is this supported? The same string works OK in the vanilla mongo client. How can I limit the result? Thanks Lars

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Group module
by Igor Olhovskiy 06 Nov '19

06 Nov '19

Hi! I'm trying to solve quite simple task and want to solve it as simple as possible. Idea is to have restriction to allow register certain username/domain only from certain subnets. So, I decided to use group and permissions modules. But stuck on group one. Code is fairly simple modparam("group", "db_url", DBURL) modparam("group", "use_domain", 1) ... if (get_user_group("From", "$avp(user_groups)")) { xlog("$fu from $si:$sp is matched group $avp(user_groups)\n"); if (allow_source_address("$avp(user_groups)")) { ...allow furter } } else { xlog("$fu from $si:$sp is not matched any groups\n"); } grp table is looks like kamctl db exec 'select * from grp' +----+-------------+-------------------+-----+---------------------+ | id | username | domain | grp | last_modified | +----+-------------+-------------------+-----+---------------------+ | 1 | 61388140395 | sip.localhost.net | 5 | 2000-01-01 00:00:01 | +----+-------------+-------------------+-----+---------------------+ But actual line of debug in logs saying this kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=720 a=26 n=get_user_group kamailio_1 | 10(18) DEBUG: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} group [re_group.c:188]: get_user_group(): getting groups for <sip:61388140395@sip.localhost.net> kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=727 a=25 n=xlog kamailio_1 | 10(18) ERROR: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} <script>: sip:61388140395@sip.localhost.net;transport=UDP from 172.21.0.1:56787 is not matched any groups What am I missing here? Or module is too old? Thanks!

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RFC: default config with jsonrpc over http instead of xmlrpc
by Daniel-Constantin Mierla 06 Nov '19

06 Nov '19

Hello, I think it was discussed during past IRC meetings or at some events across the world, with jsonrpc being these days more popular than xmlrpc, specially due to easier integration with web based apps, I am considering to replace the example of using xmlrpc over http(s) with jsonrpc in the default configuration file. It is not enabled by default for xmlrpc (requiring to define WITH_XMLRPC), so it will stay also as a define option for jsonrpc. The jsonrpcs module is already loaded for kamctl, it will require the xhttp module to be added. As for those looking for xmlrpc in the future, there is another example kamailio cfg which has the option with it: misc/examples/pkg/kamailio-oob.cfg Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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$jsonrpl(body) empty or malfunctioned during higher traffic
by Michal Škuta 05 Nov '19

05 Nov '19

Hello, Kamailio community, we are implementing evapi communication to our Kamailio server. We have implemented and tested it on our testing server. Everything was working OK. Now we are testing it on our production servers with a lot more calls and we have some problems here. Every n seconds we are requesting from Kamailio list of all dialogs. And this list of dialogs (from jsonrpc *dlg.list*) is sent to evapi connection. Here is a snippet of our *Kamailio code*: route[CGR_DLG_LIST] { xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Requested dialog list"); # CGRateS connection is still alive, set check to true $sht(cgrconn=>check) = 1; if $sht(cgrconn=>cgr) == $null { sl_send_reply("503","Charging controller unreachable"); exit; } jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.list","params":[]}'); xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Test1: $jsonrpl(body)"); evapi_relay("{\"event\":\"CGR_DLG_LIST_REPLY\", \"jsonrpl_body\":$jsonrpl(body)}"); } During a higher number of calls Kamailio is generating ERROR messages from evapi module. Here is the syslog: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [ Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_dispatch.c:764]: _evapi_relay(): cannot serialize event Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_mod.c:265]: w_evapi_relay(): failed to relay event: *We are expecting something like this when nobody is calling* Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Test1: { #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 } Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }}] (96) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:778]: _evapi_relay(): sending [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:789]: _evapi_relay(): dispatching [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) We enabled debug messages for module evapi and are trying to understand where is the problem. To make this work to this state we needed to increase the *pv_buffer* to really big size (2097152). It looks like $jsonrpl(body) is not ready and everything concated with it is not working. Any help or suggestion is welcomed. Sincerely Michal

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Module acc - How to get sip_reason/code on log_write_cdr() messages
by David VILLAUME 05 Nov '19

05 Nov '19

Hello, I'm trying to get the reason in my cdrs in log_write_cdr() messages while I see It in acc_log_request() messages. Is there a way to retrieve it in log_write_cdr() messages ? Bellow is my config : modparam("acc", "log_level", 1) modparam("acc", "log_facility", "LOG_LOCAL3") modparam("acc", "cdr_log_enable", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 2) modparam("acc", "log_missed_flag", 3) modparam("acc", "report_cancels", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_facility", "LOG_LOCAL3") modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_extra", "call_id=$ci;src_ip=$si;fU=$fU;")" David VILLAUME

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Re: [SR-Users] driver error: Can't connect to MySQL server with kamailio 5.3.x and mysql-config-groups
by Karsten Horsmann 04 Nov '19

04 Nov '19

Hello Henning, is there an planed release date for this 5.3.x version? Unfortunately without this change i cant deploy it without changing all my mysql user credentials to be sure there is no "kamailio DBURL separator" within the password field. Cheers Karsten [...] -- Mit freundlichen Grüßen *Karsten Horsmann*

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kamailio - failover setup
by Youssef Boujraf 04 Nov '19

04 Nov '19

Dear All, I have seen in the new kamailio release 5.3.0, kamailio support for haproxy protocol – both the v1 (human-readable) and v2 (binary) versions of the protocol are supported ( https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/ ) but it seems to be for Database, webserver failover not for kamailio itself. Haproxy is ONLY supporting TCP but not UDP. My question is : I am looking for failover service of kamailio servers looks like haproxy but udp & tcp. - one domain name : sip.secure.com pointing my public ip address. - two kamailio server pointing same database content (user, whitelist, ...) no asterisk or pbx. Kamailio is used Only for signaling (users, whitelist & rtp) Monitoring service : identify who is the kamailio server online, offline and the ability to force all the SIP/RTP/MANAGEMENT connection from Kamailio Srv1 to Kamailio Srv2 and vice-versa ? Thanks for your suggestions. -- Youssef

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How to use sdp_remove_media() ?
by Denys Pozniak 04 Nov '19

04 Nov '19

Hello! I'm trying to remove m=video stream from incoming SIP/SDP request in th way below: modparam("rtpengine", "read_sdp_pv", "$avp(read_sdp_pv)") modparam("rtpengine", "write_sdp_pv", "$avp(write_sdp_pv)") ... sdp_remove_media("video"); msg_apply_changes(); sdp_get("$avp(read_sdp_pv)"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| rb=/$rb/ \n"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| avp(read_sdp_pv)=/$avp(read_sdp_pv)/ \n"); rtpengine_manage("$avp(rtpengine_offer)"); set_body("$avp(write_sdp_pv)", "application/sdp"); But no luck, xlog shows m=video in $rb and $avp(rtp_sdp_read) as well. -- BR, Denys Pozniak

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Phonenum module
by Igor Olhovskiy 01 Nov '19

01 Nov '19

Hi! I'm playing around libphonenum to get validation accross e.164 numbers. So, idea is simple if (phonenum_match($fU, "src_num")) { if ($phn(src_num=>valid) != 1) { send_reply("400", "Invalid e.164 format"); xlog("L_ALERT","ALERT: FU: $fU is not e.164 number $phn(src_num=>error)\n"); exit; } } But on every (actually valid e.164 number like 380442459966 or 61388141397) I got Parsing number failed in error result. Tried both phonenum_match($fU, "src_num") and phonenum_match("$fU", "src_num") Actually phonenum_match("380442459966", "src_num") also not working Debian 10 Buster version: kamailio 5.2.5 (x86_64/linux) libphonenumber7 is already the newest version (7.1.0-5+b4).

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kamailio on kubernetes
by Gaurav Bmotra 01 Nov '19

01 Nov '19

hello everyone i m working on a project kamailio on kubernetes for high availability and auto scalability but as i m able to run it in side the cluster with using kubenetes feature but having issue to access the service form outside the cluster i m using the ingress in kubernetse to point the service its flow like that domain.com:80 ------ingress-controler--------kamailio-service:80:5060------- POD(kamailio)5060 ISSUE IS I M NOT GETING CONNECTION BITWEEN DOMAIN.COM TO POD(kamailio) is there any one ho can help me in this thank you -- *Regards:* Gaurav Kumar

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Routing Blocks - Missing in latest core wiki
by Joel Serrano 31 Oct '19

31 Oct '19

Hi, I was looking at the documentation and noticed that "Routing Blocks" section is missing in latest releases... http://www.kamailio.org/wiki/cookbooks/5.2.x/core#routing_blocks is OK But: http://www.kamailio.org/wiki/cookbooks/5.3.x/core#routing_blocks http://www.kamailio.org/wiki/cookbooks/devel/core#routing_blocks Don't exist, have they been moved elsewhere? Thanks! Joel.

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Command Line SIP client
by Gaurav Bmotra 31 Oct '19

31 Oct '19

hi i m looking for Command Line SIP client , where i can register user form CLI for testing thank you -- *Regards:* Gaurav Kumar

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How to intercept an INVITE after waking up (TSILO module) to change SDP
by Oleg Podguyko 31 Oct '19

31 Oct '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using the module rtpgngine I change SDP. Webrtc <-> SIP calls work fine. RTPengine proxies media. It is necessary to understand that when the call comes, I determine what Protocol the call came wss/tcp/udp, and then through the command "rtpengine_manage" change SDP accordingly. Now I have a new task. Wake up mobile customers through push notification. This client may be as WebRTC and also regular SIP. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { route(SEND_PUSH); xlog("L_INFO", "tsilo store\n"); ts_store(); } } switch ($var(rc)) { case 1: xlog("L_INFO", "LOCATION, Subscriber found, RC: $var(rc)\n"); route(RELAY); break; case -1: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); exit; break; case -3: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); send_reply("404", "Not Found"); exit; case -2: xlog("L_INFO", "LOCATION, Method Not Allowed, RC: $var(rc)\n"); send_reply("405", "Method Not Allowed"); exit; } } If a call occurs when the called client is registered, then we go to the route(RELAY), which creates a branch in which I change the SDP. Everything works. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (!t_relay()) { sl_reply_error(); } exit; } And already in MANAGE BRANCH the necessary SDP is put. If the called party is not registered. In the beginning everything goes well. I find this caller through a separate server. Send push notification. The client successfully registers on kamailio. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if(isflagset(FLT_SRC_WSS)) { setbflag(FLB_DST_WS); } if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu") exit; } Next I better show a piece of the log: окт 31 14:55:08 /usr/sbin/kamailio[17488]: exec: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=429 a=26 n=ts_append окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tsilo [ts_append.c:72]: ts_append(): transaction 8800:338205132 found for 1234567890, going to append branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:255]: lookup_helper(): contact for [1234567890] found by address окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:456]: lookup_helper(): instance is окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:77]: t_append_branches(): transaction 8800:338205132 in status 0 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:99]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (1) outgoing branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/msg_translator.c:162]: check_via_address(): (50.7.93.27, uas7i396ceb5.invalid, 0) окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f] with ruid [uloc-5dbacb09-4450-7] окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:163]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (0) outgoing branches after clear_branches() окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (17488 (21)), reusing fd окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2507]: tcpconn_do_send(): after real write: c= 0x7fb39dde4c68 n=3131 fd=12 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2508]: tcpconn_do_send(): buf= Thus, the transaction wakes up and the initial INVITE is sent to the new registered client without any SDP change! Which eventually leads to 415 error. Now the question itself. Tell me! As me at this moment intercept this INVITE, to it has passed through in advance prepared path for changing SDP. Thank you. -- Oleg Podguyko

1 0

kamailio - failover setup
by Ilie Soltanici 31 Oct '19

31 Oct '19

We are using keepalived with virtual IP Address for Kamailio, dispatcher module for Asterisk, 2 MySQL Databases master-master replicated and 2 Servers with ProxySQL. So far so good.

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[abalashov@evaristesys.com: binrpc reply sizes]
by Alex Balashov 31 Oct '19

31 Oct '19

Hi, What is the relationship between the `ctl` modparams `binrpc_buffer_size`, `binrpc_max_body_size`, and `binrpc_struct_max_body_size`? More especially, how interdependent are they? Is there a set of circumstances in which one would wish for the structure buffer to be smaller than the reply body buffer? What about the internal buffer? Is it used only for some kind of intermediate formatting, or does it also hold the entire reply in some aspect? The reasons for asking are predictable: RPC command 'dlg.list' throws out "reply too big" on some occasions, and rather than simply increasing all of these drastically, I would like to learn more about which ones it makes sense to increase, and in what proportion with respect to each other. I also think this idea should be more clearly developed in the documentation. Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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setbflag and isbflagset usage.
by Володимир Іванець 30 Oct '19

30 Oct '19

Hello! I'm using branch flags but not sure if the behavior is correct. There is a logic in branch route that decides if branch flag must be set and does it with this command: *setbflag(FLB_FLAG, $T_branch_idx)*. Then I'm checking if the flag is set with *if* statement later in the configuration file for requests and in onreply route for replies: *if (isbflagset(FLB_FLAG, $T_branch_idx))*. The thing is that *if* statement always returns true. The only way I can fix this behavior is to use *resetbflag* at the beginning of branch route: *resetbflag(FLB_SIPTOPUB, $T_branch_idx)*. Thank you! Regards, Volodya Ivanets.

1 0

strange conditional "if ($au!=$tU)" with previous exit
by PICCORO McKAY Lenz 30 Oct '19

30 Oct '19

i have this in asterisk integration how to, and i noted the "exit" before the "if($au!=$tU)" .. i dont understan the conditional and the exit there! please can someon xplain me that!? # authenticate the REGISTER requests (uncomment to enable auth) #!ifdef WITH_ASTERISK if (!www_authorize("$td", "sipusers")) #!else if (!www_authorize("$td", "subscriber")) #!endif { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } i investigate at the kamailio cgf documentation and there's no clear topic related! http://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#tu_-_to_uri Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

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Re: [SR-Users] [sr-dev] Kamailio as SBC for Microsoft Teams
by Henning Westerholt 30 Oct '19

30 Oct '19

Hello Rhys, I already replied to you yesterday, maybe it got into Spam on your side. :-) Quote below: > thanks. About the presence integration question - according to my research currently the only possibility would be to use the > existing Skype APIs and then use this (per interop-mode) also in MS Teams. > > Microsoft announced the availability of a API to set the presence status also for Teams, this is now scheduled for Q1 2020. There > are other limitations which also needs to be solved from MS and they are also scheduled for Q1 2020. > Feel free to contact me directly if you are interested in this topics. Cheers, Henning Am 29.10.19 um 08:51 schrieb Rhys Hanrahan: Hi Henning, This is a great article, thanks for sharing. I'm wondering if you've experimented with the possibility of integration of presence/dialog information with teams as well? Pretty much every integration guide I've seen on MS's UC solutions, including the previous ones (Lync, Communicator, SFB ) always seem limited to integrating calling via trunks. To me, the integration is not complete unless you can integrate presence things like: * If a user is on the phone then their status should be "busy" on Teams. When off call, set back to Available. * If a user is on a call in teams, our PBX should consider their PBX extension as "in use". Not sure it's possible, because I haven't seen anything of the sort being done yet, but I'm curious if this is something you looked into at all? Thanks. Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 10 307 Pitt St, Sydney NSW 2000 [http://quintus.nexusone.com.au/~rhys/nexus1-email-sig.jpg] ________________________________ From: sr-users <sr-users-bounces(a)lists.kamailio.org><mailto:sr-users-bounces@lists.kamailio.org> on behalf of Henning Westerholt <hw(a)skalatan.de><mailto:hw@skalatan.de> Sent: Tuesday, 29 October 2019 3:21 AM To: Kamailio (SER) - Users Mailing List Cc: Kamailio (SER) - Development Mailing List Subject: [SR-Users] Kamailio as SBC for Microsoft Teams Hello, I have created a “How-To” blog post on using Kamailio as “session border controller” for Microsoft Teams Direct Routing: https://skalatan.de/en/blog/kamailio-sbc-teams<https://skalatan.de/de/blog/kamailio-sbc-teams> You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

2 2

Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded
by fosdem-rtc-admin＠freertc.org 28 Oct '19

28 Oct '19

FOSDEM - Real Time Communications devroom CfP ============================================= Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2020 takes place 1-2 February 2020 in Brussels, Belgium. This document contains information about: - Real-Time Communications developer room (devroom) and lounge - speaking opportunities - volunteering in the devroom and lounge - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities) - the Planet aggregation sites for RTC blogs **NEW:** Save yourself entering Free-RTC events and CFP deadlines into your calendar and task list, follow our iCalendar feed: https://freertc.org/events.ics Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom and Real-Time lounge are about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers and participants for the tables in the Real-Time lounge.** The devroom is only on Sunday, 2nd of February 2020. The lounge will be present for both days. To discuss the devroom and lounge, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 15th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM20/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". Other devrooms and lightning talks: some speakers may find their topic is in the scope of more than one devroom. It is encouraged to apply to more than one devroom and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. Here you can find the [full list of devrooms](https://www.fosdem.org/2020/schedule/tracks/) and here you can apply for a [lightning talk](https://fosdem.org/submit). ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ----------------- To make the devroom and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - Organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 1 February - Participation in the Real-Time lounge - Circulating this Call for Participation to other mailing lists Social events and dinners ------------------------- The traditional FOSDEM beer night occurs on Friday, 31st of January. On Saturday night, there are usually dinners associated with each of the devrooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss) for further details about the Saturday night dinner options and how you can register for a seat. Related events around FOSDEM ---------------------------- As per usual, the [XMPP Summit](https://wiki.xmpp.org/web/Conferences/Summit_24) is happening ahead of FOSDEM. This time it will take place on the 30th and 31st of January also in Brussels. All details are available on the [summit website](https://wiki.xmpp.org/web/Conferences/Summit_24). Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net - (Español) https://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

1 0

Rewrite contact address on 200 OK?
by Oded Arbel 28 Oct '19

28 Oct '19

Hi, we are having a problem with a network change use case. It goes on something like this: 1. Kamailio receives INVITE from caller. 2. Kamailio sends INVITE to callee MUA. 3. MUA response with 100, then 183. 4. MUA then loses network connectivity and re-establishes access with another IP address. 5. MUA sends 200 OK from the new IP address, but with `Contact` still set as the old address (which I think it should, because you aren't allowed to change contact in the middle of dialog?) 6. Kamailio forwards 200 OK to caller and receives ACK. 7. Kamailio sends ACK to old IP address of MUA. The last step is obviously a problem as the MUA will not be able to see the ACK. I'm trying to see how to resolve this issue. I can capture the "$si" and "$sp" of the new address in the reply route (I also tried to get the MUA to send OPTIONS immediately after reestablishing connectivity, and capture "$si"/"$sp" there, which also works), but I can't figure out how to update the destination of the ACK. I've tried running `fix_nated_contact()` as well as trying to write into various pseudo variables - in either the OPTIONS request handler or the 200 reply handler - which either errors or does nothing. Any suggestions? -- Oded Arbel

2 4

Kamailio to capture mirrored traffic
by Igor Olhovskiy 28 Oct '19

28 Oct '19

Hi! I'm trying to get Kamailio working as a traffic capture on a same machine with other PBX software installed. Actually, traffic is mirrored with iptables -A PREROUTING -t mangle -i eth0 -p udp --dport 5060 -j TEE --gateway 127.0.0.2 (https://link.getmailspring.com/link/AB5F9D36-533D-4A52-ADE3-FB76B813163C@ge…) iptables -t nat -A PREROUTING -d 127.0.0.2 -p udp --dport 5060 -j DNAT --to 127.0.0.1:5062 (https://link.getmailspring.com/link/AB5F9D36-533D-4A52-ADE3-FB76B813163C@ge…) Kamailio request route is super simple request_route { xlog("L_ALERT", "[SIP-PACKET] Got packet [F=$fu R=$ru D=$du M=$rm IP=($si:$sp $Ri:$Rp) ID=$ci]\n"); drop; } I was trying to get Kamailio just listen on interface 127.0.0.1:5062, but no luck listen=udp:127.0.0.1:5062 Next was to use sipcapture module with following parameters loadmodule "sipcapture.so" modparam("sipcapture", "db_url", "text:///tmp/") modparam("sipcapture", "raw_socket_listen", "127.0.0.1:5060-5062") modparam("sipcapture", "raw_interface", "lo") modparam("sipcapture", "promiscious_on", 1) Also no luck. Means Kamailio can't see packets, but I see em with wireshark on lo interface. What is best way to get it working? Or I'm missing something? Thanks!

5 7

Cancel / OK handling
by Oz Mortimer 26 Oct '19

26 Oct '19

Hi, I have an issue where a client is sending cancel after the call has been answered but before the client received the 200OK, I know they should then respond with a BYE based on the 200 final result, but they don’t (Meaning the call / billing stays up).. Is there a way I can timeout a call/force BYE if ACK is not received to the 200 OK? I know technically this is the clients issue but they can’t / won’t fix Thanks is advance Oz.

1 0

No audio after hold and unhold event
by arish haque 25 Oct '19

25 Oct '19

Hi all, No Audio flows between the two parties when one side resumes the call after putting on hold. Configuration - WebRTC <--> Kamailio <--> Asterisk Below is the warning I'm getting in asterisk console WARNING[32388]: chan_sip.c:10425 process_sdp: Declining non-primary audio stream: audio 10596 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112 113 101 Is this a codec issue or something else. Thanks, Arish

1 0

Kamailio 3.3.7
by Brandon Armstead 24 Oct '19

24 Oct '19

Has anyone seen this error while compiling 3.3.7 raw_sock.c:431: undefined reference to `udpv4_vhdr_sum' system: Linux 4.9.0-11-amd64 #1 SMP Debian 4.9.189-3 (2019-09-02) x86_64 GNU/Linux lsmod: Module Size Used by bonding 147456 0 intel_rapl 20480 0 x86_pkg_temp_thermal 16384 0 intel_powerclamp 16384 0 coretemp 16384 0 kvm_intel 200704 0 kvm 598016 1 kvm_intel irqbypass 16384 1 kvm ttm 98304 0 drm_kms_helper 155648 0 crct10dif_pclmul 16384 0 crc32_pclmul 16384 0 joydev 20480 0 evdev 24576 3 ghash_clmulni_intel 16384 0 drm 360448 2 ttm,drm_kms_helper i2c_algo_bit 16384 0 iTCO_wdt 16384 0 shpchp 36864 0 ipmi_si 57344 0 iTCO_vendor_support 16384 1 iTCO_wdt sg 32768 0 pcspkr 16384 0 mei_me 36864 0 ipmi_msghandler 49152 1 ipmi_si mei 102400 1 mei_me video 40960 0 button 16384 0 acpi_power_meter 20480 0 acpi_pad 24576 0 ib_iser 49152 0 rdma_cm 53248 1 ib_iser iw_cm 49152 1 rdma_cm ib_cm 45056 1 rdma_cm ib_core 208896 4 ib_iser,ib_cm,rdma_cm,iw_cm configfs 40960 2 rdma_cm iscsi_tcp 20480 0 libiscsi_tcp 24576 1 iscsi_tcp libiscsi 53248 3 ib_iser,libiscsi_tcp,iscsi_tcp scsi_transport_iscsi 102400 4 ib_iser,libiscsi,iscsi_tcp ip_tables 24576 0 x_tables 36864 1 ip_tables autofs4 40960 2 ext4 589824 2 crc16 16384 1 ext4 jbd2 106496 1 ext4 crc32c_generic 16384 0 fscrypto 28672 1 ext4 ecb 16384 0 mbcache 16384 3 ext4 sd_mod 49152 4 hid_generic 16384 0 usbhid 53248 0 hid 122880 2 hid_generic,usbhid crc32c_intel 24576 2 aesni_intel 167936 0 aes_x86_64 20480 1 aesni_intel glue_helper 16384 1 aesni_intel lrw 16384 1 aesni_intel gf128mul 16384 1 lrw ablk_helper 16384 1 aesni_intel cryptd 24576 3 ablk_helper,ghash_clmulni_intel,aesni_intel ahci 40960 3 xhci_pci 16384 0 libahci 32768 1 ahci ixgbe 294912 0 xhci_hcd 188416 1 xhci_pci i2c_i801 24576 0 i2c_smbus 16384 1 i2c_i801 dca 16384 1 ixgbe ptp 20480 1 ixgbe libata 249856 2 ahci,libahci pps_core 16384 1 ptp mdio 16384 1 ixgbe usbcore 253952 3 usbhid,xhci_pci,xhci_hcd scsi_mod 225280 7 ib_iser,sd_mod,libiscsi,libata,scsi_transport_iscsi,iscsi_tcp,sg usb_common 16384 1 usbcore fan 16384 0 thermal 20480 0 I believe I've crossed this issue once before and it was a simple apt-get package to include compile options: make include_modules="mi_fifo db_mysql tm dialog sl tmx kex rr maxfwd textops textopsx nathelper acc avpops xlog uac alias_db siputils utils usrloc registrar pv regex htable sqlops sdpops topoh ctl db_mysql mysql" prefix="/usr/local" Thanks - Brandon

3 3

environmental variables in kamailio
by Gaurav Bmotra 24 Oct '19

24 Oct '19

hi i using kamailio 5.2 is it possible to use environmental variables value in kamailio.cfg like i want to use local ip and public ip in config file listen=udp:$(local_ip):5060 advertise $(public_ip):5060 if iti is possible then how i can use this thank you -- *Regards:* Gaurav Kumar

4 3

driver error: Can't connect to MySQL server with kamailio 5.3.x and mysql-config-groups
by Karsten Horsmann 23 Oct '19

23 Oct '19

Hi List, i updated one of my testing kamailios from 5.2.x to latest 5.3.x. After restarting the process, i got an mysql error that the server "kamailio" is not found. I use the config-feature of my.cnf config-group names within the kamailio.cfg I can start my kamailio 5.3.x installation with "normal username:password@host" definitions for DBURL. Did anyone have the same issue? 0(19625) ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on 'kamailio' (4) 0(19625) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 0(19625) ERROR: dialplan [dp_db.c:179]: dp_connect_db(): unable to connect to the database 0(19625) ERROR: dialplan [dialplan.c:191]: mod_init(): could not initialize data 0(19625) ERROR: <core> [core/sr_module.c:850]: init_mod(): Error while initializing module dialplan (/usr/lib64/kamailio/modules/dialplan.so) ERROR: error while initializing modules from my kamailio.cfg i defined the DBURL like this: # Use /etc/my.cnf.d/kamailio.cnf to store credentials # within a mysql-db-group #!define DBURL "mysql://[kamailio]/kamailio" and special modules can use it like this (or via DBURL) modparam("sqlops", "sqlcon", "cb=>mysql://[kamailio]/kamailio") cat /etc/my.cnf.d/kamailio.cnf # MySQL configuration group for kamailio [kamailio] user=kamailio password="kamailiorw" host=172.20.100.22 rpm -qa | grep kam kamailio-dialplan-5.3.0-13.1.x86_64 kamailio-websocket-5.3.0-13.1.x86_64 kamailio-jansson-5.3.0-13.1.x86_64 kamailio-mysql-5.3.0-13.1.x86_64 kamailio-tls-5.3.0-13.1.x86_64 kamailio-ims-5.3.0-13.1.x86_64 kamailio-statsd-5.3.0-13.1.x86_64 kamailio-5.3.0-13.1.x86_64 kamailio-redis-5.3.0-13.1.x86_64 cat /etc/redhat-release CentOS Linux release 7.6.1810 (Core) -- Cheers *Karsten Horsmann*

3 7

MS Teams Direct routing with kamailio
by Karsten Horsmann 23 Oct '19

23 Oct '19

Hello List, i search for the right place in kamailio to manipulate the dispatcher OPTIONS packets to add an Contact header. For uac generated traffic i can use the event_route[tm:local-request] from the tm module. But how can i manipulate kamailio *self* generated OPTION pings from the dispatcher? OPTIONS request for MS Teams must contain the Contact header of such a format: Contact: <sip:SBC_FQDN:5061;transport=tls> I follow this guide (for opensips) and try to adapt that step by step to kamailio: https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ CentOS 7 / kamailio 5.2.x (and maybe soon 5.3.x) / rtpengine -- Cheers *Karsten Horsmann*

2 2

Unable to use BLF
by Cehes (Free) 23 Oct '19

23 Oct '19

Hi all, This is many days we are trying many things and reading many threads :) We also re installed a simple brand new kamailio to be sure everything is ok. In fact, with sngrep we can see "subscribe" arriving on kamailio but on syslog we have a critical error : Oct 23 10:08:42 GTO-Temp01 kamailio[2004]: 20(2027) exec: {1 29 SUBSCRIBE 3135373137353930373733323337-joash9r9lnre} *** cfgtrace:request_route=[PRESENCE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=770 a=24 n=handle_subscribe Oct 23 10:08:42 GTO-Temp01 kamailio[2004]: 20(2027) CRITICAL: {1 29 SUBSCRIBE 3135373137353930373733323337-joash9r9lnre} <core> [core/mem/q_malloc.c:494]: qm_free(): *BUG: bad pointer 0xa (out of memory block!) *called from presence: *subscribe.c*: handle_subscribe(1040) - ignoring We tried to start kamailio with many more memory (x4) and this is the same. We tried with only "presenceXXX" plugins, and also with "puaXXX" plugins below is our config file : https://gist.github.com/Dyriolon/e894b9f2a67f9dfb7117421717ad6178 Any idea of what is wrong ? Thanks!

1 0

Call Pickup and Call Parking
by Zhan Bazarov 22 Oct '19

22 Oct '19

Hello, What is the Best way to implement call parking and call pickup in setup Kamailio(active-active) and dispatcher with a few asterisk round-robin? Could someone share some best practices, please? Thank you!

1 0

mysql create user and grant privileges
by Daniel-Constantin Mierla 22 Oct '19

22 Oct '19

Hello, the mysql server version 8 is no longer supporting creating the user and granting privileges in the same statement, they have to be separate. I checked the mariadb specs and they seem to support also the separate creation of user and granting privileges in two statements. I pushed the changes in kamdbctl to master branch via commit: - https://github.com/kamailio/kamailio/commit/c0879add55bef77703f2259f224b4a1… The reason I am writing here is to see if anyone is aware of some (old) versions of mariadb or mysql servers that do not support the creation and grant in two steps. If yes, then we can add a config option in kamctlrc to do one or the other variants. But for now I did it only with the new style, as it seems to be supported everywhere by latest versions, to avoid unnecessary option to keep also the old style. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 0

[Dialog] Removing entries from dialog memory
by Julien Klingenmeyer 21 Oct '19

21 Oct '19

Hi, I am wondering if there is any way to remove an entry from the active dialog list? I mean, assuming that Kamailio did not get any BYE related to a call for some reason and that I want that call to be “killed” from dialog memory before reaching the lifetime timeout. What I already tried with no success: sending a request to Kamailio catched by a dedicated route of the script then: * using the dlg_get function (with the target call as parameter) then send_bye function (then dropping the BYE request generated by the dialog module in the onsend_route with the drop function): no success. Dialog module correctly generates BYE requests but dialog related to the call is kept in memory until its lifetime timeout (and BYE requests are not dropped in the onsend_route either). * using the dlg_get function then dlg_set_timeout to set a very low timeout value for that particular call. No successs: Trying to update a bogus dlg as critical error. Could some RPC commands or whatever help me to accomplish what I try to do? Thanks Julien

4 9

pv_buffer_size limit
by Sergiu Pojoga 20 Oct '19

20 Oct '19

Hi ppl, I'm testing this external app (cgrates, you all know it) that talks to Kamailio via evapi. Every *n* minutes the app sends an evapi request to Kamailio which returns back the list of all dialogs like so: jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.list","params":[]}'); evapi_relay("{\"event\":\"CGR_DLG_LIST_REPLY\", \"jsonrpl_body\":$jsonrpl(body)}"); The problem is that the size of *$jsonrpl(body)* PV is limited by Kamailio's *pv_buffer_size* setting. My math shows that a single dialog is about 1KiB of json data. *Problem*: Kamailio fails to start with a *pv_buffer_size *higher than 117 KiB (119808 bytes), which in turn rends the system's limit to about 117 simultaneous calls... pretty sad. Below find the error when trying to start Kamailio with higher values: 0(17080) ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7f02275c5010, 120832); Free fragment not found! 0(17080) ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f02275c5010, 120832) called from core: core/pvapi.c: pv_init_buffer(2069), module: core; Free fragment not found! 0(17080) ERROR: <core> [core/pvapi.c:2072]: pv_init_buffer(): cannot init PV print buffer slot[39] Is there a workaround for this? Thanks, --Sergiu

2 1

Get value from header and check value type
by Karsten Horsmann 18 Oct '19

18 Oct '19

Hi List, I use Kamailio 5.2.x and I want to fetch an value from an X Header and do some checks like "is not empty" and "should be an integer". AFAIK per default all values are strings with the hdr function. What are the best way for some validations within Kamailio script language? if(is_present_hf("X-someheader")) { $avp(routing_target) = $(hdr(X-someheader){s.int}); if(!ds_list_exists("$avp(routing_target)")) Cheers Karsten Horsmann

2 2

How to get the dispatcher value dynamically
by Narayan P 18 Oct '19

18 Oct '19

Hello, We are using kamailio for load balancing as stateless proxy as mentioned below. {$du = "sip:192.168.130.XXX;transport=udp"; forward();} We have set $du to the correct media server value in the dispatcher module.Can you please tell me how to get the correct dispatcher value(Media server IP) and print this into log file. I am using $ru and $du to print,but it is not giving the correct one. for example: Suppose for the first call ; $du = "sip:192.168.130.XXX:5060;transport=udp"; $du = "sip:192.168.130.XXX:5062;transport=udp"; and so on. How to fetch the correct dispatcher value dynamically. Any response will be highly appreciated. Thanks, Rudra

3 2

need different Contact with the UAC modul.
by "Hans-Jürgen Brand" 18 Oct '19

18 Oct '19

3 4

Cancel / OK handling
by Oz Mortimer 18 Oct '19

18 Oct '19

Hi, I have an issue where a client is sending cancel after the call has been answered but before the client received the 200OK, I know they should then respond with a BYE based on the 200 final result, but they don’t (Meaning the call / billing stays up).. Is there a way I can timeout a call if ACK is not received to the 200 OK? I know technically this is the clients issue but they can’t / won’t fix Thanks is advance Oz.

1 0

Changing $fU or $fn in branch_route
by Jurijs Ivolga 18 Oct '19

18 Oct '19

Hi! I have small problem. When I assign $fU or $fn in request-route and then one more time I make new assignment in branch_route. Instead of rewriting it is just concatenate needed value at the end. lets assume $avp(cli)=1234567 and $avp(cliplus)=+1234567. So if in request_route I do $fU=$avp(cli); and then in branch_route $fU=$avp(cliplus);. $fU becomes one string: "1234567+1234567". Is it a bug or I missing something? Here is config snippet: request_route { ... route(MY); ... } route[MY] { ... $avp(cli) = "1234567" $avp(cliplus) = "+"+$avp(cli); $fU=$avp(cli); # here $fU=1234567 $fn=$avp(cli); # here $fn=1234567 route(LCR_ROUTE); exit; } route[LCR_ROUTE] { if(!is_method("INVITE")) return; if (!load_gws("1",$rU,$avp(cli))) { send_reply("503", "Error loading gateways"); exit; } $var(i)=0; while($(avp(lcr_gw_uri)[$var(i)])!= $null){ xlog("L_INFO", "loaded gw_uri_avp[$var(i)]=$(avp(lcr_gw_uri)[$var(i)]) \n"); $var(i) = $var(i)+1; } if (!next_gw()) { send_reply("503", "No available gateways"); exit; } xlog("L_INFO", "request-uri $ru \n"); xlog("L_INFO", "$avp(lcr_gw_uri), $avp(lcr_id)\n"); xlog("L_INFO", "flag $avp(lcr_flag) \n"); t_set_fr(0, 4000); t_on_failure("RTF_LCR_ROUTE"); t_on_branch("BRANCH_CUST"); route(RELAY); exit; } branch_route[BRANCH_CUST] { if ( $avp(lcr_flag) == 4) { $fU=$avp(cliplus); #here $fU=1234567+1234567 $fn=$avp(cliplus); #here $fn=1234567+1234567 } } Jurijs

2 5

do MOD_<module name> keywords exist?
by Juha Heinanen 18 Oct '19

18 Oct '19

I have used in config file these kind of constructs: #!ifdef MOD_<module name> ... #!endif Is the MOD_xxx keyword automatically set when module xxx is loaded? Perhaps I have invented the whole concept in a dream? -- Juha

2 4

Failed AKAv1-MD5 authentication with Boghe
by Bao, Yin 18 Oct '19

18 Oct '19

I have a Boghe (v2.0.153.836) working fine with Kamailio with MD5 algorithm however fails to authenticate if I configure S-CSCF to use AKAv1-MD5. We have the same configuration on private user identity in FHoSS on both the succ MD5 run and failed AKAv1-MD5 run. I wonder if anyone has faced this issue before. Here are some details on message exchanges: * S-CSCF included the challenge in 401 Unauthorized – Challenging the UE: WWW-Authenticate: Digest realm="example.example.org", nonce="rzGnTBGPw3hE+mDPHrZ1PAAAAAAApAAAmfftHqYUuUA=", algorithm=AKAv1-MD5, ck="629d6d9a6a6befa509b1a9bb17a9c2a3", ik="32fcd14a102279de2e38200a2257efcb", qop="auth,auth-int"\r\n * The same header was seen when 401 Unauth travelled through I-CSCF and P-CSCF (shouldn’t P-CSCF strips out the ck and ik fields before sending to UE?) * UE responded with another register with response: Authorization: Digest username="bob(a)example.example.org",realm="example.example.org",nonce="rzGnTBGPw3hE+mDPHrZ1PAAAAAAApAAAmfftHqYUuUA=",uri="sip:example.example.org",response="6445f9df2b785eab3fa461849880b48d",algorithm=AKAv1-MD5,cnonce="34baf441e99a23e0fcb2bc001355c4bf",qop=auth-int,nc=00000001\r\n * S-CSCF failed the calculation and sent back 403 Authentication Failed I tried AKAv2-MD5 setting as default algorithm on S-CSCF and the result is the same as above. It seems that only MD5 works on my setup. What are the different configs in HSS (or other places) AKAv1-MD5 need? I checked the AMF/OP on both HSS and Boghe and they match. Boghe does not have SQN setting but I set HSS to start with 0. Any pointers are appreciated! Yin

2 1

IMC through HTTP
by Péter Barabás 18 Oct '19

18 Oct '19

Hi, I want to implement conference messaging functionality into a client. IMC module seems perfect. In our scenario, the rooms and members are managed in a different server bacause of other user management functionalities and the kamailio only uses the rooms to broadcast messages to members. Is it possible anyway to create rooms and join members from a separate server like through http protocol with XHTTP module? The idea is: 1. Create a room on server A with adding members. Server A store the room data. 2. Server A sends http message to kamailio with proper format, like POST .../createRoom with jsonBody { name: „room1” } 3. Kamailio receives the http request and process it with creating a room ’room1’. 4. Server A sends members of room like POST .../addMembers with json body { room: „room1”, members: [ username: „user1”, username: „user2” ] } 5. Kamailio receives the https request and process it with joining memers into room All commands should have the belonging http request. Is it possible? Regards, Peter

2 2

MySQL vs MariaDB support
by Sergey Safarov 18 Oct '19

18 Oct '19

At present time we create RPM packages using MySQL libs. May be need switch to MariaDB libs. What you think about it?

2 1

Need to suppress "200 cancelling" in kamailio
by amit Kumar 18 Oct '19

18 Oct '19

Hi All, As per our requirement, when A party send a cancel to B party we do not want "200 cancelling" message from Kamailio.We try to find any high-level flag to disable it. but could not able to find. Can you please help us on how to achieve this. A Party Kamailio B Party ---------------->INVITE------------------------> <---------180 Ringing<------------------------ ------------>CANCEL--------------------------> <---------200 canceling [we want to suppress this message and do not want kamialio to inform A party about progress status of cancel] <------------OK----------------------------------- Thanks, Amit

2 1

Need to suppress "200 cancelling" in kamailio
by amit Kumar 18 Oct '19

18 Oct '19

Hi All, As per our requirement, when A party send a cancel to B party we do not want "200 cancelling" message from Kamailio. We try to find any high-level flag to disable it. but could not able to find it. Can you please help us on how to achieve this. A Party Kamailio B Party ---------------->INVITE------------------------> <---------180 Ringing<------------------------ ------------>CANCEL--------------------------> <---------200 canceling [we want to suppress this message and do not want kamialio to inform A party about progress status of cancel] <------------OK----------------------------------- Thanks, Amit

1 0

Kamailio v5.3.0 Released - new major version is out
by Daniel-Constantin Mierla 17 Oct '19

17 Oct '19

Kamailio v5.3.0 is out – it comes with 6 new modules and a considerable set of improvements touching more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.3.0 brings more flexibility and optimizations for KEMI interpreters, enhancements to load balancer, dialog tracking, uac remote registration and tls with libssl 1.1.x implementations, new variables, transformations and plenty of other new features. Enjoy Kamailio v5.3.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Releasing Kamailio v5.3.0
by Daniel-Constantin Mierla 17 Oct '19

17 Oct '19

Hello, if there are no known major issues that need longer time to take care of them, in my opinion we could consider releasing the next major version 5.3.0 next week, for me Thursday, October 10 seems to work the best, still allowing a bit more than one week for testing. Any other suggestions? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

3 4

catching tls errors
by Juha Heinanen 17 Oct '19

17 Oct '19

Sometimes I see in syslog errors like this: Oct 15 16:44:57 salmon /usr/bin/sip-proxy[2064]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Is it possible somehow to catch the error in config file, for example, to figure out from which IP address the connection attempt came from? -- Juha

3 6

Kamailio weighted dispatch problems
by Jasen Hall 17 Oct '19

17 Oct '19

I am trying to change our request dispatching from round-robin to a weight-based method (ds_select_dst() algorithm 9 vs. algorithm 4) and am having a hard time getting an asymmetrical dispatch pattern enabled. In testing, I'm dispatching to two Ringswitch servers and trying to send three times the traffic to one instance. My destinations list file previously set both endpoints in the same group and simply provided the SIP URI for each. My new list file adds the attribute `weight` to each endpoint and looks like this: >1 sip:10.0.0.1:5060 weight=3 >1 sip:10.0.0.2:5060 weight=1 However, when I throw a couple thousand calls at Kamailio, I always end up with an almost perfect split of calls to both ringswitches. Regardless of the weights I choose, I never see variation in the dispatched load. Does anyone have experience with this issue and have some guidance? Thanks in advance. -- Jasen Hall | Invoca *Site Reliability Engineer* www.invoca.com [image: See the list] <https://invoca.sigstr.net/uc/5cba111d3527a200469b9afd>

3 4

IMS Kamailio Diameter Charging Server
by Jack R 14 Oct '19

14 Oct '19

Hi All, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. Referring to the link below, it mentioned about Diameter based Kamailio Charging Server. https://www.kamailio.org/events/2016-KamailioWorld/Day2/15-Carsten.Bock-Dia… Where to find it? How to install it? to make it work in Kamailio IMS set up as per below articles: https://www.kamailio.org/docs/modules/stable/modules/ims_charging.html https://kamailio.org/docs/modules/devel/modules/ims_ocs.html Regards, Jack.

2 2

Kamailio v5.2.5 Released
by Daniel-Constantin Mierla 10 Oct '19

10 Oct '19

Hello, Kamailio SIP Server v5.2.5 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.5. For more details about version 5.2.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/10/kamailio-v5-2-5-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 Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

3 3

SSL Offloading with AWS ELB or Google GSLB and Kamailio
by Joel Serrano 10 Oct '19

10 Oct '19

Hi everyone, I was giving a try to setup Kamailio with a Cloud TCP load balancer in front, taking advantage of the newly added proxy protocol compatibility and my initial tests went very well. Flow: client -> (tcp) -> load balancer -> (tcp) -> Kamailio TCP socket I then did another quick test and enabled TLS, also with good results: Flow: client -> (tls) -> load balancer -> (tls) -> Kamailio TLS socket So far so good, proxy protocol works as expected. I wanted to go one step further and see if I could somehow offload SSL operations at the load balancer level, and leave kamailio handling plain tcp. Flow: client -> (tls) -> load balancer -> (tcp) -> Kamailio TCP socket This partially worked, and before I start digging into what I have to do to get it completely working, I'd like to know if anyone already has a similar setup, or even if Kamailio is able to handle such a scenario, the reason I'm asking is because of the headers, etc. In this last scenario, I receive in a TCP socket, a request with TLS headers all over the place.. INVITE sip:14a84f2016944eb0854ef0e9b71bfa10@app.mydomain.com:60655 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.16:60717;branch=z9hG4bK.KmUpamn5P;rport From: ... To: ... CSeq: 21 INVITE Call-ID: -j1QSnam9o Max-Forwards: 70 Route: <sip:sbc-test2.mydomain.com:443;lr> <http://sbc-test2.sbx.gii.me:443;lr>/> Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 436 Contact: <sip:linphone@A.B.C.D:60717;transport=tls>;+sip.instance="<urn:uuid:fabcb441-a348-49a7-948d-72448d6840eb>" I then forward this request via UDP to subsequent proxies for further processing, on the replies, my payload information back to the client should be TLS, although sent via a TCP socket.. Is this something that will not work by design? Is there any hack I can take advantage of? The goal would be for Kamailio to handle TLS headers via TCP socket, as the client expects TLS information, but the actual traffic should go in plan TCP, and the load balancer will take care of re-encrypting before replying to the client. Any ideas/suggestions/comments? I hope this email is understandable, I find it complicated to detail the exact problem, feel free to ask any questions if you don't understand anything. Thanks, Joel.

3 5

PUBLISH ERROR
by Social Boh 10 Oct '19

10 Oct '19

Hello, Kamailio 5.2.5. I'm testing presence on my Kamailio using different Clients (Jitsi, Bria5, Linphone). when they send the PUBLISH to Kamailio, all receive the same answer: 400 Invalid Request On the LOG file: ERROR: {1 21 PUBLISH O8-BtqLnzw} presence [publish.c:421]: ki_handle_publish_uri(): No E-Tag and no body found I'm pretty sure same configuration work with 5.2.3 version. Any hint? Regards -- --- I'm SoCIaL, MayBe

2 4

centos7 rpm for evapi
by Aidar Kamalov 10 Oct '19

10 Oct '19

Hello, how can I install evapi for kamailio using rpm repo? # ls /usr/lib64/kamailio/modules | grep evapi | wc -l 0 # yum provides *evapi.so Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.docker.ru * extras: mirror.linux-ia64.org * updates: mirror.corbina.net No matches found -- Aydar A. Kamalov

2 1

Releasing Kamailio v5.2.5
by Daniel-Constantin Mierla 10 Oct '19

10 Oct '19

Hello, I am considering to release Kamailio v5.2.5 out of the latest branch 5.2 on Thursday, October 10. As usual, if you noticed any patch that should be backported or you are aware of some issue not yet reported, announce it to sr-dev or via bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 1

Testing kxlibssl prng for tls module
by Daniel-Constantin Mierla 09 Oct '19

09 Oct '19

Hello Marco, I am writing to see if you can test the kxlibssl prng that I just added for tls module. I want to see if exposes the same issue you reported in: * https://github.com/kamailio/kamailio/issues/2077 If you can't test with master branch, you need to backport two commits: * https://github.com/kamailio/kamailio/commit/99eafac2d92533ba93cd8244173aef0… * https://github.com/kamailio/kamailio/commit/a52f05087a211bfecd36300907d0bff… Then set: modparam("tls", "rand_engine", "kxlibssl") The latest branch 5.2 has the code for setting custom prng backported. The idea behind kxlibssl prng is to reuse the function of the default libssl v1.1.x prng, but guarded by a kamailio specific mutex. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

2 2

Kamailio Developers Meeting, Nov 14-15, 2019, in Dusseldorf
by Daniel-Constantin Mierla 09 Oct '19

09 Oct '19

Hello, Kamailio SIP Server project is organizing another meeting of its developers during November 14-15, 2019, hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join. Please note we have a limited capacity of 20 seats in the meeting room. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 1

Fred on FLOSS Weekly 16 Oct
by jungle boogie 09 Oct '19

09 Oct '19

Hi All, I'm happy to see friendly Fred Posner on FLOSS Weekly next Wednesday (16 October, 2019) to discuss Kamailio. If you can, tune in live and see how the interview goes. Recording is at 9:30am Pacific (-7 UTC) / 16:30 UTC. https://twit.tv/shows/floss-weekly Upcoming guests: https://docs.google.com/spreadsheets/d/1_dOHpYlzRlMb0JEzwUcAUflPz8PLq7dKkos… Thanks for being on the show, Fred! Best, j.b.

1 0

exit after www_challenge
by PICCORO McKAY Lenz 08 Oct '19

08 Oct '19

i have the code with an exit, i dont know if are correct that "exit" in that line? or not? help me please? # authenticate the REGISTER requests (uncomment to enable auth) #!ifdef WITH_ASTERISK if (!www_authorize("$td", "sipusers")) #!else if (!www_authorize("$td", "subscriber")) #!endif { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

4 4

Error parsing parameter value
by Duarte Rocha 08 Oct '19

08 Oct '19

Greetings, I'm testing some parameters actions and i found a weird behaviour This is my code for the test : $var(Parameter_List) = "blue=twelve;green=twe\"lve\";yellow=\"twelve\""; xerr("Parameter List = $var(Parameter_List)"); xerr("Value of Parameter blue = $(var(Parameter_List){param.value,blue})"); xerr("Value of Parameter green = $(var(Parameter_List){param.value,green})"); xerr("Value of Parameter yellow = $(var(Parameter_List){param.value,yellow})"); The printed logs are the following : ERROR: <script>: Parameter List = blue=twelve;green=twe"lve";yellow="twelve" ERROR: <script>: Value of Parameter blue = twelve ERROR: <script>: Value of Parameter green = twe"lve" ERROR: <script>: Value of Parameter yellow = twelve I have also tried setting the string as : $var(Parameter_List) = 'blue=twelve;green=twe"lve";yellow="twelve"'; But the results were similar The "" are not recognized in some scenarios. Is this intended behaviour or should i submit it as a bug? Best Regards, Duarte Rocha

3 8

Bad BYE request
by Duarte Rocha 08 Oct '19

08 Oct '19

Greetings, This is a repeated problem, but i have more info now. Please help understand if there is a problem on my code or if there is any kind of Kamailio bug. On the first INVITE of every call i apply uac_replace_from() and uac_replace_to() in order to normalize numeration. I have the restore_mode set as "auto". The original numeration is : To : 351250000000 From : 351211455000 The INVITE is sent to the client with To and From like this and with the Record-Route headers (Kamailio has 2 Ips and inserts 2 Record-Routes) containing vsf and vst parameters to help the future restores : To: <sip:+351250000000@111.111.111.111> From: <sip:+351211455000@111.111.111.111 >;tag=BD583030373631320511EAE6 Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> My client, upon receiving the INVITE, also does some changes in the numeration and sends a 183 reply with the following headers : Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> From: "+351211455000" <sip:I2116446I_500@111.111.111.111 >;tag=BD583030373631320511EAE6 To: <sip:+351250000000@111.111.111.111>;tag=as3e46f684 After Kamailio processes the reply, To and >From are restored to it's original values and the reply is relayed : From: <sip:351211455000@111.111.111.111 >;tag=BD583030373631320511EAE6 To: <sip:351250000000@111.111.111.111>;tag=as3e46f684 Here, the restore works perfectly. By the end of the call the client sends me a BYE request with Route, From and To like this : Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62>,<sip:212.13.40.205;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> From: <sip:+351250000000@111.111.111.111 >;tag=as3e46f684 To: "+351211455000" <sip:I2116446I_500@111.111.111.111 >;tag=BD583030373631320511EAE6 Here, the difference between 183 and BYE seems to be only in the Route\Record-Route are built (2 Headers or one header with 2 uri). Now, there are a couple of problems relaying the BYE that seem to me to be related to auto replace. This BYE message isn't logged on the onsend-route and it logs a couple of errors : ERROR: <core> [core/parser/msg_parser.c:164]: get_hdr_field(): bad to header ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [To: "+351211455000" <sip:Q4525417L_<:C212.13.38.148>;tag=3DC130303736313205201F18#015#012Call-ID: 020FAAC7] ERROR: dialog [dlg_handlers.c:666]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/ ERROR: <core> [core/parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: ["+351211455000" <sip:Q4525417L_] . Despite not being logged on onsend_route and creating some errors, the BYE is still relayed like this : From: < sip:351250000000@111.111.111.111>;tag=as3e46f684 To: "+351211455000" <sip:Q4525417L_<:C111.111.111.111>;tag=BD583030373631320511EAE6 This, naturally, generates a "400 Bad Request" reply from the next equipment. I've also logged the $tU values along the code for BYE requests and it always logs the "I2116446I_500", so i don't know if my code can be the cause of this. A couple of questions and comments - What can i do to better debug this issue? I want to be sure if the problem comes from my code of if it is a Kamailio issue. Right now i'm not sure where to look. - Can a request be sent without passing on onsend_route ? I have used this code to "catch" the BYE, which works for other BYEs - if( $rm == "BYE") {xerr("On send : To User = $tU");}

3 2

SIP-links adjacency
by Oleg Belousov 07 Oct '19

07 Oct '19

Hello. In our setup kamailio perfoms some logic depending on call direction, inbound or outbound. Call direction is identified by 2 SBC IP (IP_inbound/IP_outbound), we currently use for that 2 dispatcher groups and that is working fine. Now number of peer nodes increased from one SBC till few tens, but we still need to keep SIP-links adjacency, meaning that call which arrived from SBC_N IP_inbound need to be submitted via same SBC_N IP_outbound and vice versa. There could be few approaches to implement that requirement, I'm wondering if can use existing dispatcher function/select destination logic to support SIP-links adjacency. -- obelousov.tel

3 4

q_malloc crash on Kamailio 5.2.3 with http_async_query function
by Michal Škuta 07 Oct '19

07 Oct '19

Hello everyone, I am adding some new features to our Kamailio server, one of them is sending HTTP POST request after calls end. We created simple route like this and we are calling it from event_route[dialog_end]: route[HTTP_TEST_CALL_END] { t_newtran(); $http_req(body) = "TEST"; http_async_query("http://192.168.6.23:2080/cdr_http", "HTTP_TEST_REPLY"); } Problem is that Kamailio is crashing on line *http_async_query*, the log from crash: Oct 3 16:13:37 sip_server /opt/ipgate/kamailio/sbin/kamailio[5331]: CRITICAL: <core> [core/mem/q_malloc.c:149]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten(c0003032, abcdefed)[0x97a7ff3c:0x97a7ff58]! Memory allocator was called from http_async_client: hm_hash.c:82. Fragment marked by http_async_client: hm_hash.c:71. Exec from core/mem/q_malloc.c:384. In older Kamailio 4.4 I am using http_async_query with no problems. I opened issue on GitHub: https://github.com/kamailio/kamailio/issues/2091 I hope that is ok, more logs and core dump is in the issue. Any help or suggestion is welcomed. Sincerely Michal

2 2

using http_async_query
by David Villasmil 05 Oct '19

05 Oct '19

Hello guys, So, i'm using http_async_query to send out requests. It may happen that the request times out, in this case, i would like to try a new destination, i.e. with ds_select (NOT ds_delect_dest or ds_next, because i don't want the $du to be set, since in my case i don't actually have a SIP packet). Is is possible to get the original http request url in the http_async_query timeout route? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 1

storing vars in htable
by David Villasmil 05 Oct '19

05 Oct '19

Hello guys, I'm storing some data in an htable, like so: $var(http_query) = "/default/en_US/send.html?u=user&p=pass&l=$var(next_channel)&n=$var(to)&m=$var(message)"; then i need to store it: $sht(http_requests=>$http_req_id) = $var(http_query); But what's stored is the actual vars, not the the values, it's stored like: sh store: /default/en_US/send.html?u=user&p=pass&l=$var(next_channel)&n=$var(to)&m=$var(message) Why is that? shouldn't the vars be parsed? Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Re: [SR-Users] Call limit through shared database
by Olli Attila 04 Oct '19

04 Oct '19

Hello Henning, I tried the dialog profile sharing with dmq&dialog modules and now the syncing works between the two kam nodes. Thank you! Is there a way to keep the dialog profiles persistent (load from db if kam restarts)? I was thinking a scenario where kam1 has already the call limit maxed out and kam2 suddenly reboots and after reboot, kam2 should be aware of dialogs on kam1 to be able to limit calls again correctly. Cheers, Olli

3 3

Re: [SR-Users] Call limit through shared database
by Olli Attila 04 Oct '19

04 Oct '19

Hello, I had the following enabled in db_mode param and now it is actually working: *1 - REALTIME* - any dialog information changes will be reflected into the database immediately. Restarting either of the kam1 or kam2 proxies "on the fly" will result in retrieving dialog profiles correctly from database. Cheers, Olli

1 0

Record-Route management
by Duarte Rocha 03 Oct '19

03 Oct '19

Greetings, I have multiple IPs in my Kamailio. In some scenarios, the caller and the called connect to different IPs and in others, they use the same IP. Currently i'm running with the RR's module "enable_double_rr" parameter set so i can have one Record-Route for each IP used on the call. However, in the scenario that both caller and called have the same IP, i would like to only have on Record-Route in order to save on the pactet's size. Is this possible to have those two scenarios on the same Kamailio ? Best Regards

2 2

string manipulation
by David Villasmil 03 Oct '19

03 Oct '19

Hello guys, I'm receiving an http request via xhttp, something like: http://ip.com/request?value1=123&value2=2313&value3... Is it possible to get those values somehow? I also have a suri coming from a ds_select in _dsdst_, I need to get the host from that _dsdst_ Any way of doing this? Thanks all Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Get destination out of _dsdst_
by David Villasmil 03 Oct '19

03 Oct '19

Hello guys, I've never used "ds_select" as opposed to "ds_select_dst", and i would like to get the actual destination out of _dsdst_ I've got modparam("dispatcher", "xavp_dst", "_dsdst_") and then i'm trying to do: if(!ds_select("100", "4")) { send_reply("404", "No destination"); exit; }else{ xlog("L_ERR", "[XHTTP]: The next destination for SMS is: [$xavp(_dsdst_)]\n"); ds_mark_dst("d"); } But that's coming up with The next destination for SMS is: [<<xavp:0x7fbe87acdbf0>>] I need to get the actual ip address out of the struct. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Development is open for future release of v5.4.x series
by Daniel-Constantin Mierla 03 Oct '19

03 Oct '19

Hello, the branch 5.3 was created late last evening, therefore the master branch is open for adding new features, to be part of future release series v5.4.x. Any bug fix committed to master that applies to 5.3.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.3 or 5.2. Expect that v5.3.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.4.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime in close to the end of spring 2020 or afterwards. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 0

Branch 5.3 has been created
by Daniel-Constantin Mierla 03 Oct '19

03 Oct '19

Hello, the branch 5.3 has been created, to be used for releasing v5.3.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.3 cd kamailio-5.3 git checkout -b 5.3 origin/5.3 Pushing commits in this branch: git push origin 5.3:5.3 Note that 5.3 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. You may notice a bit of more activity on tls (some issue is under work) and siptrace (a bit of cleanup for coherence might be done) modules, but over all no new features will be pushed there. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.2 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.3 will be out, respectively Kamailio v5.3.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 0

Branching 5.3
by Daniel-Constantin Mierla 02 Oct '19

02 Oct '19

Hello, if there are no strong opinions against, then it is time to branch 5.3 on git repository. I am considering to do it by the end of the day on Wednesday, October 2, 2019. After that, the master will be again open for new features, while 5.3 branch will continue to get the only fixes and improvements to documentation and tools. 2-3 weeks later we should release v5.3.0, the first stable version in 5.3.x series. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 1

Unresolvable destination for new added users...
by Zhan Bazarov 02 Oct '19

02 Oct '19

I can't make call through kamailio between new added users. I get 478 ... But it works for older users (who was added couple weeks ago e.g.) SIP flow, location and subscriber table the same in both cases. Could you please shed any light on this issue? *2019/10/02 11:11:20.719834 10.2.1.198:5068 <http://10.2.1.198:5068> -> 10.4.1.14:50600 <http://10.4.1.14:50600>* SIP/2.0 *478* Unresolvable destination (478/SL) Via: SIP/2.0/UDP asterisk_ip:asterisk_port;branch=z9hG4bK18626d14;rport=50600;received=10.4.1.14 From: "Admin Admin" <sip:some_user@asterisk_ip:asterisk_port>;tag=as09b1f1f3 To: <sip:some_user@kamailio_ip :kamailio_port>;tag=a1384aaeda4d71b5edb13882669b943e.8468 Call-ID: *20e023254f1b3c942a61e47d63badfd6@10.4.1.14:50600 <http://20e023254f1b3c942a61e47d63badfd6@10.4.1.14:50600>* CSeq: 102 INVITE Server: MS Lync Content-Length: 0

3 5

BLF 'NOTIFY' not working
by Petarr Jones 02 Oct '19

02 Oct '19

Hi I am using Kamailio 5.2 with the following PRESENCE modules and parameters: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_reginfo.so" modparam("presence", "db_url", DBURL) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "disable_bla", 0) modparam("presence_xml", "integrated_xcap_server", 1) I have a Yealink T28P phone which has two lines/extensions successfully registered with Kamailio 5.2. When I setup the BLF light to monitor one of the extensions, I get the correct SUBSCRIBE record but the NOTIFY record is sent over and over again until it fails. ----------------------------------------------------------------------------------------------------------- NOTIFY sip:101@192.168.1.123:5062 SIP/2.0 Via: SIP/2.0/UDP 45.45.45.45;branch=z9hG4bKd16c.03b5bc97000000000000000000000000.0 To: <sip:101@45.45.45.45>;tag=1366853594 From: <sip:104@45.45.45.45>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-e6c3 CSeq: 2 NOTIFY Call-ID: 594386350(a)192.168.1.123 Content-Length: 0 User-Agent: kamailio (5.2.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:45.45.45.45:5060;transport=udp> ----------------------------------------------------------------------------------------------------------- When I register the phone with Freeswitch instead and run the same test, the NOTIFY works and I get a much longer NOTIFY message ----------------------------------------------------------------------------------------------------------- NOTIFY sip:100@192.168.1.123:5064 SIP/2.0 Via: SIP/2.0/UDP 66.66.66.66;rport;branch=z9hG4bK1mXej9r2605UB Route: <sip:55.55.55.55:5064>;transport=udp Max-Forwards: 70 From: <sip:102@66.66.66.66>;tag=xUoVghM5syHc To: "100" <sip:100@66.66.66.66>;tag=1720892352 Call-ID: 2583553127(a)192.168.1.123 CSeq: 1178415701 NOTIFY Contact: <sip:102@66.66.66.66:5060> User-Agent: FreeSWITCH-mod_sofia/1.10.1-release-12-f9990221e6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1800 Content-Type: application/dialog-info+xml Content-Length: 149 xml version="1.0"?> dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:102@66.66.66.66"> dialog-info> ----------------------------------------------------------------------------------------------------------- Should the message coming from Kamailio contain more data or is that not the issue. Should I be including other modules also? All guidance would be appreciated.

2 4

Call limit through shared database
by Olli Attila 01 Oct '19

01 Oct '19

Hello, I have 2 kam instances using one shared database and both kam servers running the same routing logic code. Kamailio servers are configured to use the same database as dialog backend. I am limiting the concurrent dialogs in the following way but it seems that kamailio is using its own memory to read profile size rather than the db. Second kam is not aware of the dialog profiles that the first kam wrote to the shared db. Is there a way to get multiple kam instances to read profile size from the same database when calling the get_profile_size function? modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) route[LIMIT_CALLS] { if (!dlg_isflagset("1")) { # $td – gateways, $fu - user if (get_profile_size("concurrent_calls", "$fu", "$avp(calls)")) { if ($avp(calls) >= 3) { xlog("L_INFO", "Concurrent calls $fu at limit"); send_reply("503", "Calls limit reached"); exit; } else { dlg_manage(); dlg_setflag("1"); set_dlg_profile("concurrent_calls", "$fu"); } } } } Cheers, Olli

5 8

Kamailio Charging
by Jack R 01 Oct '19

01 Oct '19

Hi, Does Kamailio provide some charging functionality? If yes, then how do we go about setting it up? like what module/configuration? and Database tables to see/touch? I see these links: https://www.kamailio.org/docs/modules/4.2.x/modules/ims_charging.html https://www.kamailio.org/w/2013/07/documentation-on-charging-in-ims/ Or Do we need separate software like example cgrates for charging? (also a side question do we have any other open-source like cgrates) Any information would be helpful. Thanks and Regards, Jack.

3 3

Relay Request
by Maryam Baghdadi 29 Sep '19

29 Sep '19

Hello everyone, Is it possible to relay INVITE request to another server after receiving "180 Ringing" response ??? I would like to activate CRBT(ring back tone) service and according to my research after receiving 180 ringing media playing request should be sent to a media server .

3 2

Re: [SR-Users] Relay Request
by Mojtaba 29 Sep '19

29 Sep '19

Would you please keep "Users Mailing List" <sr-users(a)lists.kamailio.org> in tracking. :) That is straight froward. There is no change in sending the request from Kamailio to Asterisk. In Asterisk you could use Progress command to support Media-Relay. In additional use Dial command (with r or m option for supporting CRBT ). With Best Regards On Sun, Sep 29, 2019 at 8:52 AM Maryam Baghdadi <maryam.baghdadi(a)gmail.com> wrote: > Hello Mojtaba, > I chose asterisk as media server and some services like voice mail > provided ... about CRBT I know asterisk have this service but I don't know > how to send this request from kamailio to asterisk.. > do you know kamailio can provide this service itself or not??? > could you pls explain more > > thanks > > > On Sat, Sep 28, 2019, 1:48 PM Mojtaba <mespio(a)gmail.com wrote: > >> Hello, Some consideration would be needed: >> 1) SS7 for PSTN subscriber to support media-relay >> 2)Progress capability for SIP subscriber. >> 3) Media server to support ring-back VAS service >> You should check enabling this service or not during MO procedure. >> With Best Regards. >> >> On Sat, Sep 28, 2019 at 1:19 PM Maryam Baghdadi < >> maryam.baghdadi(a)gmail.com> wrote: >> >>> Hello everyone, >>> >>> Is it possible to relay INVITE request to another server after receiving >>> "180 Ringing" response ??? >>> I would like to activate CRBT(ring back tone) service and according to >>> my research after receiving 180 ringing media playing request should be >>> sent to a media server . >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> -- >> --Mojtaba Esfandiari.S >> > -- --Mojtaba Esfandiari.S

1 0

tcp_accept_haproxy questions
by Grant Bagdasarian 27 Sep '19

27 Sep '19

Hello, The docs mention regular TCP connections not being accepted anymore once this is enabled. I assume UDP is unaffected and both proxyprotocol and udp may live side by side? Also, does this play nicely with WebSockets? Taking into account the Websocket connection being setup through a proxy protocol supporting proxy. Regards, Grant

1 0

wiki on $sht(htable=>key)
by Juha Heinanen 27 Sep '19

27 Sep '19

Kamailio wiki does not tell what result $sht(htable=>key) gives if the key is not found in htable. It would be nice if it did. -- Juha

1 0

KEMI dialplan dp_replace returns not implemented yet
by Grant Bagdasarian 26 Sep '19

26 Sep '19

Hello, The dp_replace KEMI function is returning the message "not implemented yet". Function call: _.dialplan.dp_replace(1, _.pv.get("$rU"), "$var(dpr)") app_python3 ¡apy_kemi.c:872¿: sr_apy_kemi_f_pv_get_mode(): pv get: $rU <core> ¡core/pvapi.c:398¿: pv_cache_lookup(): pvar ¡$rU¿ found in cache app_python3 ¡apy_kemi.c:193¿: sr_apy_kemi_exec_func_ex(): execution of method: dp_replace app_python3 ¡apy_kemi.c:338¿: sr_apy_kemi_exec_func_ex(): not implemented yet app_python3 ¡apy_kemi.c:872¿: sr_apy_kemi_f_pv_get_mode(): pv get: $var(dpr) <core> ¡core/pvapi.c:371¿: pv_cache_add(): pvar ¡$var(dpr)¿ added in cache Using Kamailio 5.2.4 Docker image. version: kamailio 5.2.4 (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 with gcc 6.3.0 Regards, Grant

2 4

Kamailio Rasberry Pi benchmark results
by Henning Westerholt 26 Sep '19

26 Sep '19

Hello, as promised here the report on my Rasberry Pi benchmarks. The numbers were quite impressive! - 3000 registration requests per second (in-memory usrloc) - 300 calls per second for a p2p call (default cfg, only SIP traffic) - about 18 MBit/s incoming and 16 MBit/s outgoing SIP peak traffic could be processed All the details about the test setup and more numbers in the blog post: https://skalatan.de/blog/kamailio-rpi-benchmark Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

Kamailio v5.1.9 Released
by Daniel-Constantin Mierla 26 Sep '19

26 Sep '19

Hello, Kamailio SIP Server v5.1.9 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.8. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.9 or to v5.2.x series. For more details about version 5.1.9 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/09/kamailio-v5-1-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.4 being released out of it. 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 Kamailio World Conference - April 27-29, 2020 -- www.kamailioworld.com

2 1

ims_ipsec_pcscf: maybe core dump due to maximum count of connections
by Valentin Christoph 26 Sep '19

26 Sep '19

Hi, Doing some tests with the module ims_ipsec_pcscf, we encountered a core dump during startup of Kamailio and would like to understand the correct behavior. We assume the ipsec_max_connections parameter was the root cause --> how many max connections have been actually tested? Can it be that ipsec_client-port and ipsec_server_port were too close together (some overlapping)? Thx Christoph

2 1

install with kazoo
by Benoit Chesneau 26 Sep '19

26 Sep '19

Hi all, Can anyone share a simple configuration to use with kazoo or a link to it? I'm struggle in having it setup without using the defaults packages. Ideally it should dispatch messages to freeswitch. Any help is welcome :) Benoît

3 2

Planning the release of v5.1.9
by Daniel-Constantin Mierla 26 Sep '19

26 Sep '19

Hello, I am considering to release v5.1.9 out of the branch 5.1 during the next week, on Tuesday (Sep 24) or Thursday (Sep 26). As usual, if you are aware of issues not reported on tracker or commits that should be backported to branch 5.1, do report them asap in order to get them a chance to be sorted out. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 2

Kamailio ARM support improved, tests
by Henning Westerholt 25 Sep '19

25 Sep '19

Hello, as part of the testing for 5.3.0 I looked into the ARM support in Kamailio. Previously the core used some deprecated assembler functions for atomic operations (locking etc..). There was also no native support for the ARM v7 architecture available. After several fixes and extensions both topics should be now much improved. I did today a lot of tests on Rasberry Pi and also some benchmarking (will send another e-mail when the report is ready). If you use Kamailio on another embedded ARM (>= v6) system it would be great if you also give it a try. Use git master revision ccc0eb6d12a1d (or newer) for your tests. Please report eventual issues to the sr-dev list. Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

Development for v5.3.x is frozen
by Daniel-Constantin Mierla 25 Sep '19

25 Sep '19

Hello, the formal notification that the development for the next major version 5.3.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.3 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, pull requests can still be done, but they will be merged after branching 5.3. Can still be done commits with documentation improvements, enhancements to related tool (e.g., kamctl, kamcmd) and exporting missing KEMI functions to the modules already exporting most of the other functions (done for previous releases, because likely that was a fail to notice, all functions should be exported for a module at the same time). Once the branch 5.3 is created, new features can be pushed again to master branch as usual. From that moment, the v5.3.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.3 is collected at: * https://www.kamailio.org/wiki/features/new-in-5.3.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.1.x-to-5.3.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.2.x to 5.3.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 8

Extract user from AOR
by Grant Bagdasarian 25 Sep '19

25 Sep '19

Hello, I've been trying to extract the user part from an AoR using transformations in Python (KEMI), but without success. The reg_fetch_contacts function has been called successfully before trying to access the AoR. None of the below statements work for me. They all return None. user = _.pv.get("$(ulc(profile=>aor){uri.user})") user = _.pv.get("$(ulc(profile=>aor){tobody.user})") or _.pv.sets("$var(aor)", _.pv.get("$(ulc(profile=>aor))")) user = _.pv.get("$(var(aor){tobody.user})") user = _.pv.get("$(var(aor){uri.user})") I'm not sure what's causing the None result. Could it be the curly braces being used for string interpolation in Python? Or are these transformations not able to extract the value since the AoR has the following format: user@domain? If the same can be accomplished in another way, please let me know. Regards, Grant

2 1

Question
by Shahdee Nelson 25 Sep '19

25 Sep '19

With your software would I be able to create, and modify VOIP lines? As in saying would I be able to provide VOIP numbers that can make outbound calls to USA from USA?

2 1

Problem restoring To and From (uac_replace)
by Duarte Rocha 24 Sep '19

24 Sep '19

Greetings, I'm getting some errors with the auto restore of From and To from UAC Module. After the restore, the URI has lots garbage and it's not like the original. I think the problem lies on the way the client sends the Route Header to me, but i'm not sure. When Kamailio sends the INVITE Record-Route are like this : Record-Route: <sip:xxx.xxx.xxx.xxx;................> Record-Route: <sip:aaa.bbb.aaa.bbb;................> However, when my client sends a BYE to finish the call, the Route are on this mode : Route: <sip:aaa.bbb.aaa.bbb;................>,<sip:xxx.xxx.xxx.xxx;................> The module's documentation says this : "One uses the rr module and appends to the Record-Route header a parameter containing some strings from which the original and new URI can be computed. The problem with this mode is that it relies on the fact the parties will send the Route exactly as it was received. In case there is a change, the resulting URIs will not be correct." Can this change by the client be the cause of this issue? Is there anything i can do to avoid this? Best Regards, Duarte Rocha

1 0

Can I load balance Kamailio with Kamailio?
by Ne Mo 24 Sep '19

24 Sep '19

3 2

Parsing json file on kamailio.cfg file
by Melek Oktay 24 Sep '19

24 Sep '19

Hi, I am trying to monitor *uac *channels status periodically in kamailio config file. I call uac.reg_dump rpc command and result will be inside in $var(uac_channel) variable jsonrpc_exec('{"jsonrpc": "2.0", "method": "uac.reg_dump"}'); $var(uac_channel) = $jsonrpl(body); it returns me json formatted result like below, it has a several channels info and their status *flags* {"jsonrpc":"2.0", "result":[ {"l_uuid":"YY","l_username":"YY","l_domain":"YY","r_username":"YY","., " *flags*":16 }, {"l_uuid":"XX","l_username":"XX","l_domain":"XX","r_username":"XX",,... " *flags*":1 }}]} I could not parse this result on kamailio.cfg file ! I want to learn channel that has a 16 value in flag attribute for example !! Would you help me to figure out how I am gonna parse this json file please ?

2 2

qm_find_free() Free fragment not found, called from xcap_server, no more pkg
by Cody Herzog 23 Sep '19

23 Sep '19

Hello. Recently, for the first time, we experienced an apparent memory issue in our production environment. Here's an example of the relevant log messages: Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: <core> [core/mem/q_malloc.c:286]: qm_find_free(): qm_find_free(0x7fd8bbcfc010, 134648); Free fragment not found! Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fd8bbcfc010, 134648) called from xcap_server: xcap_server.c: ki_xcaps_put(549), module: xcap_server; Free fragment not found! Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: xcap_server [xcap_server.c:552]: ki_xcaps_put(): no more pkg Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2252]: ERROR: app_perl [kamailioxs.xs:487]: XS_Kamailio_log(): 500 Server error The failed operation was the XCAP server module trying to generate a very large RLS contact list for a user. The issue only impacted users with very large lists, as though a large contiguous block of memory could not be found, whereas other smaller allocations continued to work fine. We believe the requested allocation was around 112 KB in size. The server had been up for 14 days. We were able to work around the issue temporarily by just restarting the Kamailio service. It's unusual, because our production server is often up for months, and we've never seen this issue before. The load on production is increasing slowly due to an increased concurrent user count, so that might be related. Before restarting the service on production, we captured the output of the following commands: kamctl stats shmem kamcmd mod.stats all shm kamcmd pkg.stats kamcmd mod.stats all pkg Here's the shared mem output: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 27240", "shmem:free_size = 447203296", "shmem:max_used_size = 116175576", "shmem:real_used_size = 89667616", "shmem:total_size = 536870912", "shmem:used_size = 68824240" ], "id": 6934 } Here's the pkg output for the particular PID which was throwing the errors: { entry: 34 pid: 2302 rank: 14 used: 2415864 free: 4949688 real_used: 3438920 total_size: 8388608 total_frags: 1951 } We didn't see anything obvious in the stats output which explains the issue. We've been trying to reproduce the issue in a dev environment using simulated higher than production load running for many days, but so far we've had no luck. We've been monitoring memory stats over time, but we don't see any obvious leaks or issues. We've searched various past threads, but didn't find any obvious answers. Here are some of the documents and the threads we've been reading: https://www.kamailio.org/wiki/tutorials/troubleshooting/memory https://www.kamailio.org/wiki/cookbooks/3.3.x/core#mem_join https://sr-users.sip-router.narkive.com/3TEDs3ga/tcp-free-fragment-not-found https://lists.kamailio.org/pipermail/sr-users/2012-June/073552.html https://lists.kamailio.org/pipermail/sr-users/2017-February/096132.html https://lists.kamailio.org/pipermail/sr-users/2017-September/098607.html https://github.com/kamailio/kamailio/issues/1001 https://lists.kamailio.org/pipermail/sr-users/2016-April/092592.html https://lists.kamailio.org/pipermail/sr-users/2010-July/064832.html Regarding our Kamailio version and build options, here's the output of 'kamailio -v': ---- version: kamailio 5.1.0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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_LISTEN 16, 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 20:07:31 Jan 4 2018 with gcc 5.4.0 ---- We are running on a 64-bit Ubuntu Server virtual machine. Any help would be greatly appreciated. Thanks very much.

3 6

Using static entries with usrloc
by George Diamantopoulos 23 Sep '19

23 Sep '19

Hello, Normally I wouldn't be asking this but I was surprised to find no reference to anyone doing this online. I need to have a few AoRs with static entries in kami's location table, which will always be reachable at their contacts without ever having to register with Kamailio. Am I right in thinking that merely INSERTing respective rows in kamailio's location DB backend will do the trick? Is there anything I need to worry about, like contacts being periodically purged? If yes, how do I prevent that from happening? And will kamailio always query the DB when performing location lookup or do I need to restart upon inserting new records (to purge caches, for example)? Lastly, I'm not sure how to treat some fields present in usrloc tables like ruid, expires (can I set this to NULL?), callid and cseq (which is irrelevant since there's no REGISTER), flags, cflags and methods. Thanks, George

5 7

xtrange error but those rules that already work, some bug?
by PICCORO McKAY Lenz 23 Sep '19

23 Sep '19

i have many xtrange errors in the log.. but the code seems as i guess completed each rule without error. In the kamailio i have that error by example: Sep 23 13:38:29 ser-server-02 kamailio[2847]: 4(2855) ERROR: *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio-devel.cfg] l=692 a=25 n=save that line represents a supposed error in the following code, BUT that code seem work.. kamailio it’s working perfectly.. so i’m worry about it, what it happened? # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } why that happened? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

kamailio run in docker not working
by Gaurav Bmotra 23 Sep '19

23 Sep '19

hi i m trying to run kamailio in docker this kamailio/kamailio-ci:master-alpine but inside the container kamailio not running if i rung this / # /etc/init.d/kamailio status sh: /etc/init.d/kamailio: not found plz help me thanks -- *Regards:* Gaurav Kumar

2 3

rate-limiting enabled on kamailio.org
by Henning Westerholt 22 Sep '19

22 Sep '19

Hello, just to inform you that rate-limiting on kamailio.org web server has been enabled. The motivation is to throttle robots and other automated services to not overload the server. If you notice any new errors in accessing the service (like page not found, or a 429 Too Many Requests) let us know. Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

TLS in-dialog set_forward_no_connect()and upstream TLS LCR gateway
by Anthony Joseph Messina 21 Sep '19

21 Sep '19

In preparation for the 5.3 release, I've been testing the following configuration change for TCP/TLS connections: https://github.com/kamailio/kamailio/commit/ 8bba208fe6ae7ccb4c92362b8c33f1530b9f56da route[REQINIT] { # no connect for sending replies set_reply_no_connect(); if(has_totag()) { # no connect for requests within dialog set_forward_no_connect(); } This change creates issues when a UAC TLS INVITE routes to an upstream gateway using TLS to port 5061 (via the LCR module). Kamailio sends the initial outbound TLS connection from a local ephemeral port. The TCPOPS tcp_keepalive_enable function issues keepalives from the local ephemeral port to the gateway port 5061: https://kamailio.org/docs/modules/stable/modules/ tcpops#tcpops.f.tcp_keepalive_enable Even so, the TLS connection eventually times out, after which in-dialog requests from the UAC are no longer able to reach the upstream gateway. ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failed WARNING: tm [t_fwd.c:1570]: t_send_branch(): sending request on branch 0 failed ERROR: sl [sl_funcs.c:372]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) I figure I must be doing something wrong with my TCPOPS here. Is a TLS connection to an upstream gateway supposed to be maintained throughout the duration of a call? -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 6

Info: initialization of libssl 1.1.+ with PTHREAD_PROCESS_SHARED attribute for pthread mutex and rwlock
by Daniel-Constantin Mierla 21 Sep '19

21 Sep '19

Hello, after couple of months waiting for feedback about the workaround with the preloaded shared library to set the PTHREAD_PROCESS_SHARED attribute for pthread mutex/rwlock in order to avoid deadlocks and as it seems to be the solution, I just pushed a few commits to move that code in the core of kamailio. It this way the initialization with the PTHREAD_PROCESS_SHARED attribute is done by default when libssl v1.1 is detected at compile time, without the need of using the approach with a preloaded library. The shared library is still kept in the tls module: * https://github.com/kamailio/kamailio/tree/master/src/modules/tls/utils/open… Probably is going to be removed in the near future. I am asking the community members that use Kamailio on systems with libssl v1.1+ to test the master branch and give feedback. The master branch is anyhow in testing phase for preparing the next major release v5.3.0, so it will help for that as well. If no issue is going to be reported soon, the commits will be backported to stable branches, to re-enable smooth use of Kamailio with libssl v1.1+. When libssl v1.1+ is detected, the core links with pthread, but that was already present in the Makefiles.defs for different compile options. The pthread more or less a default library on linuxes/unixes, so this should not introduce any major dependency to the Kamailio core. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 21

Possible locking issue in usrloc module in file udomain.c
by Rupesh Kumar 18 Sep '19

18 Sep '19

Hi , Currently I am running kamailio 4.2 and now upgrading to kamailio 5.2 and was just going thru the changes. In the file modules/usrloc/udomain.c inside the function free_udomain( ) , it appears that we are not protecting via lock_ulslot and unlock_ulslot doing free. Could it cause some race condition ? May be its intentional but going thru the github history its there since day 1. I checked all branches in kamailio 4.x and we have the protection. In 4.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { lock_ulslot(_d, i); deinit_slot(_d->table + i); unlock_ulslot(_d, i); } shm_free(_d->table); } shm_free(_d); } In 5.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { deinit_slot(_d->table + i); } shm_free(_d->table); } shm_free(_d); }

2 2

How to distinguish is from Asterisk or from the client
by przeqpiciel 18 Sep '19

18 Sep '19

Hi, I looking information how to distinguish whether the packet is directed to the Astersik (my platform) or to the customer (softphone / sip client ).

5 5

NAT problem
by Arnout Van Den Kieboom 18 Sep '19

18 Sep '19

Hi, First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one. I started using kamailio in a test environment about a week ago. for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio. I have installed rtpproxy and it works nicely. There is just one situation where it fails: Carrier --> FW (NAT) Kamailio --> asterisk There is only one way audio (from asterisk to carrier) but not from carrier to asterisk. I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy... Thanks Pemp

4 7

app_lua: invalid Lua environment attributes or parameters
by Yuriy Gorlichenko 18 Sep '19

18 Sep '19

Hi! Im using kamailio 5.1.8 with app_lua and getting this message during startup of kamailio. Looks like I forgot to add something at the module setup, but cant figure out what is wrong. My setup of app_lua is: modparam("app_lua", "load", "/etc/kamailio/extended/main.lua") modparam("app_lua","reload",1) modparam("app_lua","register","textops") app_lua [app_lua_sr.c:1599]: sr_kemi_lua_exec_func_ex(): invalid Lua environment attributes or parameters starting kamailio service with kamailio -f /etc/kamailio/extended/main.lua Also here is little bit confusing me that in case of use "KSR" everywhere I'm getting message about "sr"... Thx in advice

2 5

dedicated bearer - Kamailio settings
by Jack R 18 Sep '19

18 Sep '19

Hi, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu. I have done the attached (image) settings to create dedicated bearer in nextepc as advised by this link: https://github.com/open5gs/nextepc/issues/180 Do I need to do some setting in Kamailio server side (maybe in pcscf cfg files etc.to take some information from PCRF) to make dedicated bearer work end to end or that is all (just configure the new apn in the phone) rest will be taken care itself? Thanks and Regards, Jack.

2 1

core dump if silo table > ~5000 records
by David Villasmil 18 Sep '19

18 Sep '19

Hello guys, i've got a kamailo: version: kamailio 5.0.0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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_LISTEN 16, 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 19:50:58 Mar 28 2017 with gcc 4.9.2 which was core dumping every 20 seconds and we had no idea why. We emptied out the silo table, which had: mysql> select count(*) from silobackup; +----------+ | count(*) | +----------+ | 5240 | +----------+ records. Once we truncated it, kamailio started working again. Is there a limit to how many records msilo can handle? I will look into the data in the table, but in case there's a limit, i'd like to know. core available on request. Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Kamailio+B2BUA on IPv4-only and NAT64 Apple app store tests
by Anthony Alba 17 Sep '19

17 Sep '19

Hi folks, I'm wondering what you do to pass Apple app store tests (IPv6-only NAT64/DNS64 scenario) with the iOS app likely to put IPv6 addresses in the SDP, when Kamailio and your B2BUA(Asterisk, FreeSWITCH,...) are IPv4-only endpoints. By "pass" I mean media flowing + SIP working - not sure the app store gatekeepers would actually test media, but it would be nice to have a robust solution. Strictly speaking, this is not a Kamailio's problem but a media SDP/<your-rtp-proxy-of-choice> problem. But since everything is so closely intertwined I'm wondering where you choose to the location the "solution". I have noticed that if Kamailio is IPv4-only and it "sees" IPv6-addresses(Contact, Via headers??) it will try to resolve the IPv6-address as a DNS name! The Wikipedia article on NAT64 mentions 464XLAT(RFC 6877) and Port Control Protocol (RFC 7225), but those seem to be deep rabbit-holes... Cheers Anthony Alba

4 5

Kamailio merchandising shop
by Henning Westerholt 16 Sep '19

16 Sep '19

Hello, there is now a Kamailio merchandising shop available: https://skalatan.de/merchandising/ You find there many options e.g. different Kamailio T-Shirts, Hoodies, Bags, Caps, Mugs, etc.. It provides shipping in many countries world-wide (based from Europe and also USA). Feedback via (direct) e-mail welcome. Best regards, Henning

1 1

3GPP compliance
by Tihomir Culjaga 15 Sep '19

15 Sep '19

Hi, is there any 3GPP compliance list for Kamailio IMS ? Regards, Tihomir.

2 1

RUN kamailio in kubernetes
by Gaurav Bmotra 14 Sep '19

14 Sep '19

hi i want to run kamailio in kubernetes deployment and i want to run this kamailio/kamailio-ci:5.2 image in to the kubernetse .. can anyone help thank you -- *Regards:* Gaurav Kumar

5 6

Problem with incoming audio from Asterisk
by przeqpiciel 14 Sep '19

14 Sep '19

Hi, After a month of fight with SIP signalling I made it but now I have problem with incoming audio from Asterisk. Kamailio and Asterisk are their own public IPs I dont use rtpproxy. When Asterisk sends rtp I'm able to see RTP on my router but it not going to my computer. Maybe somebody have idea why ?

2 1

Using $shv within current call only
by Zhan Bazarov 13 Sep '19

13 Sep '19

Hello, we ran into a trouble with sharing variable into reply route... We are going to match asterisk's uniqueid with rtpengine records like this: route[from_asterisk]{ $var(UniqueId)=$hdr(X-UniqueId); $var(rtp_flags) = $var(rtp_flags) + " label={uid:$var(UniqueId),stream:outgoing} record-call=on"; rtpengine_manage($var(rtp_flags)); } it nicely work for outgoing request, but doesn't for reply.... reply_route[from_pstn]{ rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing"); } and it doesn't work... The one possible way what we found its using $shv instead of $var/$avp... But its unsecure way, cuz it may confus our bussines logic... Do you have any idea how to implement this, how we can assign $shv within current call only? Many thanks!

3 2

Issue with <null> and 0 when comparing vars
by Joel Serrano 13 Sep '19

13 Sep '19

Hello, I'm running into what I believe a weird situation when comparing <null> to 0. I have the following lines: xlog("L_NOTICE", "DEBUG: sel(contact.expires)=$sel(contact.expires) - hdr(Expires)=$hdr(Expires)\n"); if ($hdr(Expires)==0 || $sel(contact.expires)==0) { xlog("L_NOTICE", "[end] removed user location\n"); } else { xlog("L_NOTICE", "[end] saved user location\n"); } I registered with 3 devices, one sends expires as header, another as contact param, and the last uses both header AND contact param. My surprise was this: 1- Both header and contact param: Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=600 Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: [end] saved user location 2- Only contact param: Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=<null> Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: [end] removed user location 2- Only header: Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: DEBUG: sel(contact.expires)=<null> - hdr(Expires)=3600 Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: [end] removed user location I can workaround the header by using is_present_hf() before comparing, I can also probably parse the $ct to see if it has some Expires in it, but I really want to know if my understanding is incorrect and I'm doing something wrong or if this is a bug? Thanks! Joel.

2 2

Testing server up with SIP OPTIONS/netcat
by Andrew White 13 Sep '19

13 Sep '19

Hi all, As part of a custom monitoring solution, I’m writing a simple script that sends an OPTIONS via netcat to a Kamailio server. However the server doesn’t appear to like it. Here’s the text - options.sip: OPTIONS sip:10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.20:5060 From: <sip:monitoring@10.0.0.20> To: sip:monitoring@10.0.0.10 Call-ID: monitoring(a)10.0.0.20 CSeq: 1 OPTIONS I’m running the following command from the 10.0.0.20 machine: nc -u 10.0.0.10 5060 < options.sip And I get the following error in the Kamailio logs: ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] WARNING: <core> [core/receive.c:319]: receive_msg(): parsing relevant headers failed ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: pv [pv_core.c:1893]: pv_get_hdr(): error parsing headers ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} maxfwd [mf_funcs.c:51]: is_maxfwd_present(): parsing MAX_FORWARD header failed! ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/msg_translator.c:2361]: build_res_buf_from_sip_req(): alas, parse_headers failed As far as I can see, the message itself is valid. It appears parse_cseq() is expecting a higher cseq, or more headers? I’ve looked at another OPTIONS message we get regularly from one of our upstream, and it’s almost identical to this one. Can anyone point me the right way here? Thanks! Andrew

3 2

Presence and BLF
by Bill Neely 12 Sep '19

12 Sep '19

Hi all: I am trying to set up presence and BLF notifies on Kamailo 5.2.4 I am following the instructions laid out in http://kb.asipto.com/kamailio:presence:k43-blf Don't know if these instructions are valid, but I am definitely not getting any light action (using Grandstream 2130 and Grandstream Wav for testing.) Entries in active_watchers appear to be correct. Here is my cfg file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v5.1 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" #!define WITH_DEBUG #!define WITH_NAT #!define WITH_AUTH #!define WITH_MYSQL #!define WITH_USRLOCDB #!define WITH_PRESENCE ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_DLG 9 #!define FLT_DLGINFO 10 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=3 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* 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:62.151.xx.xx:50060 # allow http requests listen=tcp:127.0.0.1:8080 tcp_accept_no_cl=yes #!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" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "104.192.xx.xx" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_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" loadmodule "presence_dialoginfo.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.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 # load the module loadmodule "xhttp.so" loadmodule "json.so" loadmodule "pua_json.so" loadmodule "dialog.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- 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) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:62.151.xx.xx:50060" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 2) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) # ----- presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_single_dialog", 0) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) #!endif # ----- dialog params ----- modparam("dialog", "db_url", DBURL) modparam("dialog", "enable_stats", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### 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 { xlog("L_ALERT","ALERT: REQUEST ROUTE ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 $rm from $fu (IP:$si:$sp) $du\nMESSAGE BUFFER:\n$mb\n\n\nRU:$ru\n"); $var(i)=0; while($var(i)<$branch(count)) { xlog("$(branch(uri)[$var(i)])\n"); $var(i) = $var(i) + 1; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle registrations route(REGISTRAR); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); xlog("L_ALERT","ALERT: line 492******************************** $rm from $fu (IP:$si:$sp)\n"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_ALERT","ALERT: line 494******************************** $rm from $fu (IP:$si:$sp)\n"); record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } event_route[xhttp:request] { if ($hu != "/presence/") { xlog("L_INFO", "invalid request [$hu]"); xhttp_reply("404", "Not Found", "text/html", ""); return; } $var(call-id) = $(rb{json.parse,Call-ID}); xlog("L_INFO", "$var(call-id)|log|[$hu] payload $rb"); if ($(rb{json.parse,Event-Package}) == "dialog") { xlog("L_INFO", "$var(call-id)|log|received $(rb{json.parse,Event-Package}) update for $(rb{json.parse,From}) $rb"); pua_json_publish($rb); } xhttp_reply("200", "OK", "text/html", "$rb"); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); dlg_manage(); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($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); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { xlog("ALERT:NATMANAGE********************* (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("corwie"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains # Routing to foreign domains route[SIPOUT] { xlog("ALERT: SIPOUT (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } route[TOAST] { rewritehostport("45.33.xx.xx:5060"); $fu = "sip:7028743833@62.151.182.83:5012"; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 2

Kamailio IMS setup - user online - unable to call
by Jack R 12 Sep '19

12 Sep '19

Hi All, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. Users alice and bob are online however unable to call. When we check the logs the FHoSS is hunting for values in sh_subscription table for alice / bob user. Can you please share: 1. A sample database table values to be inserted across all tables in FHoSS helpful for the call and 2, Maybe, if possible, some description to understand table structures/ importance of fields of FHoSS. Thanks and Regards, Jack.

2 1

Help with "routines:ssl3_read_bytes:sslv3 alert bad certificate" kamailio TLS error
by david＠aslo.us 12 Sep '19

12 Sep '19

Hello everyone, I am trying to configure TLS in kamailio (5.2.4) following this guide: [ http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates ]( http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates ) Modules: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ALIASDB #!define WITH_IMC #!define WITH_TLS When i try to connect via command line, this is the result (just including relevant parts): $ openssl s_client -connect 192.X.X.X:5061 -tls1 CONNECTED(00000003) depth=1 C = XX, ST = XXXX, L = XXXXXX, O = XXX CA, CN = XXX CA verify error:num=19:self signed certificate in certificate chain verify return:0 --- No client certificate CA names sent --- SSL handshake has read 2550 bytes and written 336 bytes --- --- Start Time: 1567787935 Timeout : 7200 (sec) Verify return code: 19 (self signed certificate in certificate chain) --- read:errno=0 Now, when I setup my clients, they connect to the server, but they can't send messages or make calls. This is the TLS startup LOG: Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_mod.c:372]: mod_init(): With ECDH-Support! Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_mod.c:375]: mod_init(): With Diffie Hellman Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: CRITICAL: tls [tls_init.c:671]: init_tls_h(): installed openssl library version is too different from the library the kamailio tls module was compiled with: installed "OpenSSL 1.1.1 11 Sep 2018" (0x1010100f), compiled "OpenSSL 1.1.0k 28 May 2019" (0x101000bf).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: WARNING: tls [tls_init.c:680]: init_tls_h(): tls_force_run turned on, ignoring openssl version mismatch Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: WARNING: tls [tls_init.c:778]: init_tls_h(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 12582912 and 6291456 bytes Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 12582912 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 6291456 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [main.c:2669]: main(): processes (at least): 24 - shm size: 67108864 - pkg size: 8388608 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/certs/192.X.X.X/cert.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='/etc/certs/demoCA/cert.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/certs/192.X.X.X/key.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: NOTICE: tls [tls_domain.c:1087]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:707]: set_verification(): TLSs<default>: No client certificate required and no checks performed Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=12 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:710]: set_verification(): TLSc<default>: Server MAY present invalid certificate Sep 6 16:41:58 aslo-kamailio /usr/sbin/kamailio[5862]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/5862 Sep 6 16:41:58 aslo-kamailio /usr/sbin/kamailio[5866]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) This is my tls.cfg file: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/certs/192.X.X.X/key.pem certificate = /etc/certs/192.X.X.X/cert.pem ca_list = /etc/certs/demoCA/cert.pem #crl = /etc/kamailio/tls/crl.pem # --- # This is the default client domain profile. # Settings in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] method = TLSv1 verify_certificate = no require_certificate = no These are the relevant parts of my kamailio.cfg: # alias="sip.mydomain.com" alias=192.X.X.X:5060 alias=192.X.X.X:5061 /* 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:192.X.X.X:5060 listen=tcp:192.X.X.X:5060 listen=tls:192.X.X.X:5061 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif These are the errors that show up everytime i try to connect with a client: Sep 6 16:53:42 aslo-kamailio /usr/sbin/kamailio[5870]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:42 aslo-kamailio /usr/sbin/kamailio[5870]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Sep 6 16:53:43 aslo-kamailio /usr/sbin/kamailio[5874]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:43 aslo-kamailio /usr/sbin/kamailio[5874]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Sep 6 16:53:44 aslo-kamailio /usr/sbin/kamailio[5875]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:44 aslo-kamailio /usr/sbin/kamailio[5875]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Any help would be greatly appreciated. Regards.

2 1

Kamailio status get Active (Exited)
by Gaurav Bmotra 11 Sep '19

11 Sep '19

hi i m using kamailio 5.1, with -m 512 -M 8 it stop registering user SIP anf in status it shows ------------------------------ ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Sat 2019-09-07 11:05:16 UTC; 3 days ago Docs: man:systemd-sysv-generator(8) Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17708]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17677]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17714]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17685]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17681]: {1 1094155754 NOTIFY TwGwmzx4wINnraJs1Gh3Kg..} INFO: <core> [main.c:819]: sig_usr(): s… received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17704]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17689]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17693]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17691]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17674]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Hint: Some lines were ellipsized, use -l to show in full. --------------------------------------------------------------------- plz help thank you -- *Regards:* Gaurav Kumar

2 1

Can't connect to db
by Andrew Chen 10 Sep '19

10 Sep '19

Hi all, I just installed kamailio 5.2.4 on two separate aws instances in two separate regions. Both regions uses the same configuration template and share the same db in AWS RDS. The issue I see is this: - SJO location was able to launch fine without any issues. - ASH location could not start up kamailio process due to db connection failure: Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8043]: ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Too many connections Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8043]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Too many connections Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: carrierroute [db_carrierroute.c:141]: carrierroute_db_open(): can't connect to database. Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: <core> [core/sr_module.c:847]: init_mod_child(): error while initializing module carrierroute (/usr/lib/x86_64-linux-gnu/kamailio/modules/carrierroute.so) (idx: 30 rank: 18 desc: [tcp receiver (generic) child=9]) - In ASH instance, I can manually mysql to the RDS host using the credentials in the config file. - I checked db user permissions and it's opened for the subnet in our ASH location (which is also proven using manual connection). So I don't get why one works and the other doesn't Thoughts? -- Andy Chen Sr. Telephony Lead Engineer -- *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

LCR lcr_id short or int?
by Grant Bagdasarian 10 Sep '19

10 Sep '19

Hello, The database tables definition for 5.2.x mentions the lcr_id parameter being of unsigned short type, while in the code I only see definitions of type int. Is it safe to assume the parameter is of type int, so the database column can also be set to an integer type? Regards, Grant

2 7

Share variable into reply route.
by Zhan Bazarov 09 Sep '19

09 Sep '19

Hello, we ran into a trouble with sharing variable into reply route... We are going to match asterisk's uniqueid with rtpengine records like this: route[from_asterisk]{ $var(UniqueId)=$hdr(X-UniqueId); $var(rtp_flags) = $var(rtp_flags) + " label={uid:$var(UniqueId),stream:outgoing} record-call=on"; rtpengine_manage($var(rtp_flags)); } reply_route[from_pstn]{ rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing"); } and it doesn't work... The one possible way what we found its using $shv instead of $var... But its unsecure way, cuz it may confus our bussines logic... Do you have any idea how to implement this? Many thanks!

3 2

How to install kamailio symbols for debugging
by YASIN CANER 09 Sep '19

09 Sep '19

Hello, I want to debug kamailio with gdb or try to get core.dump file but i failed. GDB wants to kamailio symbols. How to install them? on Ubuntu - raspberry PI Best regards. Yasin CANER

4 3

Error Users - 604 Does not exist anywhere - HSS User Unknown
by Jack R 08 Sep '19

08 Sep '19

Hi, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. 1. 2 Sip client registers successfully (shows Online) as 'bob(a)net1.test' and 'alice(a)net1.test' 2. When we check server logs: Invite request shows 'bob@IP address' i.e. ' bob(a)10.0.0.10' 3. However, in HSS it is registered as 'bob(a)net1.test'. 4. When the request from SCSCF goes to ISCSF it gives Error: Users - 604 Does not exist anywhere - HSS User Unknown. As it compares 'bob(a)10.0.0.10' with 'bob(a)net1.test'. Please help and advise how everywhere (SCSCF, PCSCF) during query/comparisons it can take 'bob(a)net1.test' and not 'bob(a)10.0.0.10' Thanks and Regards, Jack.

1 1

Kamailio - PfSense Firewall - inbound/outbound rules
by Youssef Boujraf 06 Sep '19

06 Sep '19

Dear All, I am evaluating the kamailio proxy 5.2 server with rtpproxy on an internal network. Everything are ok. Now, I would like to evalute Kamailio proxy from outside (internet) but I have some difficulties to setup my PfSense firewall to enable inbound & outbound rules, network settings and kamailio.cfg. The issues are : Setup correctly Network files : * /etc/hostname 127.0.0.1 localhost ip_private codecsip.mydomain codecsip ip_public codecsip.mydomain codecsip * /etc/hosts file codesip * dns * /etc/resolv.conf nameserver 8.8.8.8Setup /etc/kamailio/kamailio.cfg :#!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS /* 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="codecsip.mydomain" #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") Setup etc/default/rtpproxy : CONTROL_SOCK=udp:127.0.0.1:7722 EXTRA_OPTS="-l <IP-address>" Where <IP-address> is the external IP address of your host. Regarding the firewall rules, it will be great to identify which rules I need to create : Inbound and outbound rules and NAT 1:1 ? If anyone have any information it could be great Best regards, Youssef

2 1

Kamailio HA
by Christian Attanasio 06 Sep '19

06 Sep '19

Hi everyone I use kamailio 5.1 with rtpengine and I would like to set up a simple warm standby or hot standby system for my sip server. on the web I found various discussions and posts that led me to deepen them the dispatcher, dmq, dmq_usrloc, db_cluster and p_usrloc modules. now, though, I'm a little confused: what approach do you recommend for this system? I thought of using two identical servers: Kam01 10.10.10.2 Kam02 10.10.10.3 Hostname cluster.mydomain.com each with its own mariadb and implementing data replication (with the dmq module? Or db_cluster and sqlops?). in this scenario is it necessary to also use the dispatcher module or implement a high available system with keepalives and floating ip? thanks, Christian

1 0

Kamalio & Asterisk integration erroneous behavior, setup issue
by Aristeidis Tsitras 05 Sep '19

05 Sep '19

new to the area and trying to setup Kamailio with Asterisk in a single machine. All users will register to Kamailio's port and in case of need for media, it will be forwarded to Asterisk, that is my intention. All of my work is based on the following link https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. Here is what i have done: - Debian 8, 64 bit machine with mysql and odbc - *root@kamast: ~ $ lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 8.11 (jessie) Release: 8.11 Codename: jessie root@kamast: ~ $ uname -a Linux kamast 3.16.0-10-amd64 #1 SMP Debian 3.16.72-1 (2019-08-13) x86_64 GNU/Linux root@kamast: ~ $ * - Kamailio 5.2 installed from Kamailio's deb repository - Asterisk 13LTS installed from source - Used the same passwords such as kamailiorw and asterisk_password, since this is a test system, for proof of concept. I did import to the mysql>asterisk database 3 users 2200, 2201 and 2202. Then created in sip.conf the same 3 users with the same credentials. Then on 3 PCs i used softphones (Jitsi, Zoiper) and registered each account to a softphone. Problems: - Cannot see the users in the Asterisk's cli, sip show peers - I can see users only in Kamailio with kamctl ul show - A call between the extensions goes to voicemail. It never reaches the other destination eg 2200 calls 2201 and in Asterisk's console i am getting a message that 2201 is absent and it goes to voicemail. The same with any other extension. Attached you can find: 1. Kamailio.cfg 2. Asterisk's sip.conf 3. Asterisk's extension.conf 4. The import that i have done to mysql for the user creation. I would appreciate if someone could point me to the error and help me fix it please?

3 7

$Au being set to $fu instead of $fU
by Joel Serrano 05 Sep '19

05 Sep '19

Hi, Testing on latest master deb nightly build from last night I've noticed that when $au is not available, $Au is being set to $fu (uri) instead of $fU (username). Docs: https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables ... $Au - Acc username: username for accounting purposes. It's a selective pseudo variable (inherited from acc module). It returns auth username ($au) if exists or From username ($fU) otherwise. ... I have the following log in several places in a test box: xlog("L_NOTICE", "[DEBUG] Au=$Au au=$au aU=$aU fU=$fU fu=$fu - \n"); And when I run through them (on both request and reply routes) I get the following: Aug 8 14:37:43 csbc01 csbc[9478]: NOTICE: {1 21564 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9479]: NOTICE: {1 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=1002366 aU=1002366 fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9484]: NOTICE: {2 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Those 3 logs belong to the same flow from different places in the config script. I would expect $Au to never have a "@some.domain" in it, so I assume in some place the replacement is not being done with $fU but with $fu. @Devs: I'm happy to create a GH issue for tracking etc if required, please let me know! Cheers, Joel.

2 3

How to detect miss call and accomplished call
by Anuran Barman 05 Sep '19

05 Sep '19

Hi, I want to show call logs in my SIP app. How can I do this with Kamailio? I want to do it from the *kamailio.cfg *file. Is it possible to do it from where? I guess Dialog module should be used for this, but can not figure out how to do it. I want minimalist feature like who call whom with timestamp, thats all. How can I do this?

3 9

Kamailio-IMS system
by Hạnh Nguyễn 05 Sep '19

05 Sep '19

Dear Mrs/Mss, I am Hanh, I am from Viet Nam. I am writting this email because I have an issue when I download The log-file and the config-files on website https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/. When I click on https://www.dropbox.com/s/ehkzi9pbhlp7njc/Kamailio-IMS%20config-files.zip?d… to download The log-file which I really need on this time, the page displays 404 Error - Link not found. So I can not download. I hope I will receive a new link from you soon. I look forward to hearing from you. Sincerely, Hanh Nguyen

2 1

Planning the next major release - v5.3.0
by Daniel-Constantin Mierla 04 Sep '19

04 Sep '19

Hello, it's about the time to set the milestones to the next major release - v5.3.0. There are 6 new modules and many other enhancements to existing components in git master branch, also during the last irc devel meeting it was proposed to freeze after summer holidays. At this moment I would propose Wednesday, September 4, 2019 to freeze the development. Then the release should be about one month later, after testing interval. In case there are people willing a bit more time for development, we can freeze one week later, September 11, or another date -- your suggestions here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 5

Using Kamailio IMS for making a call
by supreeth herle 04 Sep '19

04 Sep '19

Hello Everyone, I have successfully setup a VM in OpenStack with Kamailio IMS, rtpengine and FoHSS using the instructions mentioned in the guide which i wrote (attached to this mail), all running in the machine. I am using the existing alice and bob IMPUs created in FoHSS. But, when I try to call each other the call doesnt go through and I observe in the wireshark traces that S-CSCF is replying to I-CSCF that "403, Forbidden - Domain not served". Can somebody please tell me do I need any additional components to make a call or am i missing some configuration? It would be great if someone can help me troubleshooting this issue. I have attached the pcap file for your reference. Many thanks in advance. Regards, Supreeth

1 0

Kamailio stops processing requests over UDP
by Juha Heinanen 04 Sep '19

04 Sep '19

I'm returning to an old issue, where Kamailio stops processing incoming requests. Now it turned out that only requests over UDP are not processed. Requests over TCP work fine. Debug log doesn't show anything about the requests over UDP, but wireshark sees them coming. core.ps shows that main process - attendant and all udp receiver processes are alive. Any ideas what could cause Kamailio (5.2 version) to stop processing requests over UDP? What additional info should I try to get? -- Juha

4 21

Hello Group - I have a "challenge"
by Kevin Greene 04 Sep '19

04 Sep '19

I recently did the upgrade from v4.4 to 5.2 (via Yum). When starting Kamailio, I get these three errors. Not sure if they are related or not but I do not think so. Have spent a lot of time searching on Google for the answer and have not found the solution. [root@ids kamailio]# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : INFO: started (pid: 44608) [root@ids kamailio]# tail -f /var/log/sipserver.log Sep 4 00:39:06 ids /usr/sbin/kamailio[44608]: ERROR: <core> [core/pvapi.c:1119]: pv_parse_spec2(): wrong char [t/116] in [$mt] at [2 (0)] Sep 4 00:39:06 ids /usr/sbin/kamailio[44608]: ERROR: <core> [core/dprint.c:451]: log_prefix_init(): wrong format[{$mt $hdr(CSeq) $ci} ] Sep 4 00:39:06 ids /usr/sbin/kamailio[44626]: ERROR: <core> [core/receive.c:319]: receive_msg(): no request_route {...} and no other config routing engine registered The last error is repeated many times (each time a remote user tries to connect I think). I can send any additional files if needed. Thanks. I am burning out on this upgrade. Kevin

2 1

Unexpected BYEs from dialog module
by Alex Balashov 03 Sep '19

03 Sep '19

Hi, I am running 4.3.5:1b0c0a, which I am aware is an EOL'd release train, and have a problem with the dialog module I am baffled by. On many calls - I can't find any correlation to particular kinds of endpoints - I see spoofed BYEs come out of Kamailio after a minute and a half, as if the call had hit a dialog timeout or the dialog module's dead peer detection (OPTIONS pinging) kicked it out. The thing is, neither of those explanations seem to be borne out by the parameters of the calls under investigation. The dialog timeout on these calls is set to 7200, and the dialog keepalives aren't enabled in either direction. If they were, it'd be logged. Calls that are terminated in this manner also see the following log message: WARNING: dialog [dlg_req_within.c:214]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7f9997317e48 [3390:9644] with clid '1996679936_133218050(a)x.x.x.x' and tags '2bb17663-co4006-INS001' 'as7925780d' But I don't think this is unusual for a dialog-spoofed BYE; presumably this is due to a 200 OK for the BYE from both ends. So anyway, I am trying to track down anything else that could inadvertently cause dialog to hang up calls relatively quickly in that release, or inadvertently set the dialog timeout parameter to something lower than is apparent from the logging and the config. If anyone has any pointers, that would not go unappreciated! Many thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

crash/mem leaks on low memory
by Gaurav Bmotra 03 Sep '19

03 Sep '19

hi i m using kamailio 5.1 on Ubuntu 18.04 i m geting this error in logs and it stop registration form soft-phone plz help me -------------------------------------------------------------------------------------------------------- WARNING: tls [tls_init.c:778]: init_tls_h(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory th -------------------------------------------------------------------------------------------------------- kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Tue 2019-09-03 12:09:19 UTC; 26min ago Docs: man:systemd-sysv-generator(8) Process: 27237 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 7077888 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [main.c:2636]: main(): processes (at least): 27 - shm size: 536870912 - pkg size: 8388608 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: ERROR: dispatcher [dispatcher.c:813]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: ERROR: <core> [core/tcp_main.c:2850]: tcp_init(): bind(c, 0x7f0b41b001f4, 16) on 172.31.35.49:5091 : Address already in use Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized -- thank you *Regards:* Gaurav Kumar

3 3

evapi mismatch version
by bill＠novatrope.us 02 Sep '19

02 Sep '19

Hi All Installed Kam 5.2.4, but now getting version mismatch on evapi. ERROR: <core> [core/sr_module.c:348]: version_control(): module version mismatch for /usr/lib64/kamailio/modules/evapi.so; core: kamailio 5.2.4 (x86_64/linux); module: kamailio 5.1.6 (x86_64/linux) Anybody know how to get the correct evapi version. I am running centos 7 Bill

3 2

Issue with uac_replace
by David Costa 01 Sep '19

01 Sep '19

Greetings, I have a Kamailio proxy with uac_replace for To and From Headers. So far everything has worked correctly. However, I now have a client who encodes my Calling Number too and I'm getting some issues in the restore mechanics. I have the restore setup as "auto" and it's info is stored in "vsf" and "vst" parameters of Record-Route. On replies sent by my client I have the following Record-Route and From Header (Note : My Kamailio has two IPs which makes it insert two RR headers ): Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> From: "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this case the restore is done correctly When the client sends me a BYE request I have the following Route Header and To Header Route : <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2>,<sip:xxx.xxx.xxx. xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_o=128_11_Y;vsf=AA AAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcH wIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> To : "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this scenario, the To after restore is : To :"+351411110097" <sip:Q4525417L_<>Gxxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4, which throws an internal error on Kamailio about bad URI and also makes our MediaGateway reply with a "Bad Request". Can this error be caused by the restore of the uac_replace mechanics? Both Record Route URIs are on different headers while the both Route URIs are on the same Route header. Can this be the reason? If you need more information please let me know. Best Regards David Costa

2 1

Re: [SR-Users] controlling presence with new module `pua_json`
by Henning Westerholt 30 Aug '19

30 Aug '19

Hello Bill, (please keep the list in CC) yes, this seems to be the case. Just looked briefly to the code, it does an assignment in some cases. E.g. if no Presentity* data is given it will just use the From* information. And then it should later output an error from pua_json_update_presentity(..) if there is not the necessary data. Do you see any error message like this? Another idea - does it works if you add more "missing" data in the curl requests? Cheers, Henning Am 28.08.19 um 21:16 schrieb bill(a)novatrope.us: > Henning: Good point. > > I do see a number of errors in the log. It looks like the json routine > is looking for data that is not in the http query as specified by the > author. > > > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Name: [update] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Package: [dialog] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-User] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-User: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-Realm] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-Realm: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): From: > [sip:1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-User: [1020101] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Realm: [<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To: > [sip:1020108@<IP of server>:50060] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > To-User: [1020108] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Call-ID: [1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Direction] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Direction: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): State: > [Confirmed] > 17(125579) DEBUG: presence [event_list.c:348]: search_event(): start > event= [dialog/5] > > On 8/28/19 12:00 PM, Henning Westerholt wrote: >> Hello Bill, >> >> just giving some generic debug pointers here: >> >> - do you see any errors in the kamailio log file? >> >> - do you see any SIP traffic (e.g. by observing with ngrep, wireshark >> etc..) to the phone after your curl command? >> >> Cheers, >> >> Henning >> >> Am 28.08.19 um 19:04 schrieb bill(a)novatrope.us: >>> Hi all: >>> >>> I am trying to use the procedure for setting BLF lights described in >>> https://blog.voipxswitch.com/2018/02/22/kamailio-controlling-presence-with-… >>> >>> >>> WIthout success. >>> >>> My configuration is 1 Grandstream GXP2130 phone behind a NAT on a >>> public IP. (IPP in example below) >>> >>> Kamailio is running on public IP (IPK in example) listening on port >>> 50060 >>> >>> The command I am sending: >>> >>> curl -d >>> '{"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"}' >>> >>> http://localhost:8080/presence/ >>> {"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"} >>> >>> >>> >>> >>> Kamailio appears to be processing the command OK, so I think I am just >>> not setting it up properly> >>> >>> Anybody having any luck with this procedure, let me know how you are >>> getting it to work. >>> >>> >>> Bill >>> >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 1

Info: Upgrading kamcli to work with Python3
by Daniel-Constantin Mierla 30 Aug '19

30 Aug '19

Hello, with the perspective that Python2 is going to be removed from the major Linux distributes starting with the beginning of next year, the work to upgrade kamcli to run with Python3 has been started. >From now on, master branch of kamcli is expected to work only with Python3: * https://github.com/kamailio/kamcli There is still work to be done to upgrade all the code, but any use of it should be done with Python3 and eventual issues to be filed on bug tracker. Of course, pull requests are welcome as always! Who wants to use the version for Python2 has to use the code from git branch v1.2-python2. That branch is probably not going to receive any new feature, it will be kept for use with older systems (as well as a running option during the upgrade period to Python3). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

IMC module not sending messages to members of a conference room
by david＠aslo.us 29 Aug '19

29 Aug '19

2 1

Kamailio 5.0.8 | authentification issue only with TCP/TLS
by Laurent Schweizer 29 Aug '19

29 Aug '19

Dear all, I have a kamailio running in version 5.0.8 and since fee weeks we have an issue with different users connected in TCP or TLS, sometimes authorization like for REGISTER are rejected and after a moment (can be few minute or hours) it work again and of course no change was done in the password .... We see this issue with different device, snom swyx, ... and on UDP we have no issue. I can see that when the Register is rejected it's with the error -2, so wrong password... # Authentication route route[AUTH] { if (is_method("REGISTER")) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { switch($retcode) { case -1: sl_send_reply("503","Service not available"); exit; case -2: xlog("L_WARN", "auth error -2 username $au - src ip: $si \n"); auth_challenge("$fd", "0"); exit; I have attached an example of a trace where we can see a first REGISTER accepted and less than 2 minutes after a new one is rejected. ( in between they is a REGISTER without any Authorization header) Any idea ? BR Laurent

2 3

Connect Kamailio to ITSPs?
by Aingaran Thirunadarajah 29 Aug '19

29 Aug '19

Hello, Newbie here! Have Kamailio set up on a 512MB VPS. Created a couple of users, registered with Zoiper and can call between them. I am looking for documentation to connect the server to other VoIP providers (Signalwire, Anveo, VOIP.ms etc.) to make & receive calls from the external network (DIDs) / PSTN Can someone point me in the right direction? Thanks in advance.

2 1

Define properly IP in record-route header
by przeqpiciel 29 Aug '19

29 Aug '19

Hi, I have some issue and dont have enought knowledge to fix it. I have simple infrastructure. Internet -> router (NAT) -> kamailio -> asterisk Kamailio and Asterisk are inside the network and Kamailio places his own private address in the "record-route" headers instead of who's public. There is configuration to said to him what IP is proper ?

3 8

kamailio+asterisk VoiceMail implementation
by Zhan Bazarov 28 Aug '19

28 Aug '19

Hello, what's best practice for VoiceMail implementation in Kamailio+Asterisk's scheme? should I forward register into asterisk? Or I can keep register on kamailio? Could you, please, shed any lights on this

2 1

Disable GW's on runtime
by Duarte Rocha 28 Aug '19

28 Aug '19

Greetings, I have two Kamailio's working on a HA scenario. Both Kamailio's have DB replication and each machine has one HA IP. In order to set this up, i have every GW on the DB duplicated, one with Socket IPA and another with socket IPB. I have both GWs in a active state configurated in the replicated DB. They also both have the same URI and the same group id Example : GW1 : Socket = IPA ; GroupID = 1 ; Uri = sip:1.1.1.1:5060 GW2 : Socket = IPB ; GroupID = 1 ; Uri = sip:1.1.1.1:5060 In order to avoid errors, i want to disable on runtime the GWs that have a socket that doesn't belong, at the moment, to that machine (GW with Socket IPB on machine with IPA). To do that i would use the dispatcher.set_state, however it works with group id and address, which in this case it is the same. Is there other method to disable the GWs on runtime using other arguments like duid or socket ? I thought about using probing for this, since it will disable automatically the GWs with non-existent socket, but not all my clients support probing, so it isn't feasible. Thanks in advance

2 5

UAC ha1 password
by Alexandru Covalschi 28 Aug '19

28 Aug '19

Hello, I'm trying to make Kamailio to SIP register on a remote server. However storing plaintext password looks very unsecure. Is there a possibility to store password for uac module using ha1 hash instead of the plaintext password? I see there's a row for it in the database, but in the source code it's not used and it seems i can't neither use it in db neither set up in-memory as avp. Maybe there's a workaround to directly access the in-memory uac registration htable? ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

3 5

controlling presence with new module `pua_json`
by bill＠novatrope.us 28 Aug '19

28 Aug '19

Hi all: I am trying to use the procedure for setting BLF lights described in https://blog.voipxswitch.com/2018/02/22/kamailio-controlling-presence-with-… WIthout success. My configuration is 1 Grandstream GXP2130 phone behind a NAT on a public IP. (IPP in example below) Kamailio is running on public IP (IPK in example) listening on port 50060 The command I am sending: curl -d '{"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"}' http://localhost:8080/presence/ {"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"} Kamailio appears to be processing the command OK, so I think I am just not setting it up properly> Anybody having any luck with this procedure, let me know how you are getting it to work. Bill

2 1

[tm-local-request] usage
by Duarte Rocha 28 Aug '19

28 Aug '19

Greetings, Does "event_route[tm:local-request]" works like onsend route ? I'm having some problems and i'm using local-request to debug and i found this on the documentation : "Executed after the tm module has sent a local generated, transaction stateful request." I'm logging all the requests that go through this route. Is this really the last place before kamailio send the request? If the request is logged here, is there anything more in kamailio that can stop the request (pipelimit, etc) ? If a packet gets logged on this route does that mean that it was already sent? Best Regards, Duarte Rocha

3 4

Kamailio 4.4.7 - LCR modules
by Laura 28 Aug '19

28 Aug '19

Hi guys, on our enviromet we have 130+ gateway configured over the LCR_gw tables. when we try to reload it.. we get error.. "Aug 28 09:40:36 vm-nextaitz-02 /usr/sbin/kamailio[3800]: ERROR: lcr [lcr_mod.c:1511]: reload_tables(): too many gateways" We reached some top values.. how can we modify it ? Regards

2 2

Dispatcher - FQDN with multiple IPs
by Sergiu Pojoga 27 Aug '19

27 Aug '19

Hi ppl, The problem I'm facing is that function *ds_is_from_list* seems to return false when a gateway's address is a FQDN that resolves into multiple IP addresses (I'm talking about A records), consequently in-dialog requests like BYE and re-Invites get blocked since they don't pass the *ds_is_from_list* validation. Am I missing some parameter or something? Thanks, --Sergiu

3 3

RFC: about sl and tm local-response event routes
by Daniel-Constantin Mierla 27 Aug '19

27 Aug '19

Hello, just discovered what I consider to be an inconsistency in naming and behaviour for event_route blocks for local-request and local-response and starting a discussion here to see how to move on. The event_route[tm:local-request] is executed before sending there local generated request out (allowing also to change its content, drop, etc...). This event route is quite popular event route used when the local generated requests need to be checked or updates. The event_route[tm:local-response] is executed after the response is sent out, obviously no possibility to change anymore the content. I haven't checked the code for event_route[sl:local-response], but based on commit message should be the same. I haven't used the local-response so far at all, today after a discussion on sr-users I wanted to enable kemi callback for tm:local-response and I noticed that behaviour in the code, even I expected to be like local-request (before sending out). I am not sure how much used are the event routes for tm:local-response and sl:local-response, I haven't seen any questions about them so far on mailing lists, that's why I am asking here if would make sense to rename them like tm:local-response-sent and sl:local-response-sent to properly reflect when they are executed. I am expecting that they are very few used so far, so no big head ache with upgrades and bringing some consistency around (this change to be part of next major release). It can stay like now with proper documentation, however in the future if one want and event route for local responses before being sent out, there will be more confusion, imo ... Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

4 5

Too many hops 483 - unable to setup call
by Jack R 27 Aug '19

27 Aug '19

Hi, I installed Kamailio SIP server: Following article: https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… By following step # 12 (as per above article) when we initiate the call we get - Too many hops 483 Created two user earlier: $ kamctl add test testpasswd $ kamctl add test2 testpasswd syslog show: Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56276]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (45.45.0.16:42965) to sip:test1@139.54.102.135, 04e06b6352f7a2b8f97c1a3733b3334a(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: ims_registrar_pcscf [save.c:339]: save_pending(): Will save pending contact without security parameters Aug 27 09:37:05 localhost /usr/sbin/kamailio[56374]: ERROR: <script>: Diameter: AAR failed on subscription to signalling Server IP 139.54.102.135 where Kamailio is installed, UE IP address 45.45.0.16, 45.45.0.1 Followed this https://www.kamailio.org/wiki/tutorials/faq/main#why_the_sip_requests_are_r… Thought to test using SIPp - installed and ran default command: ./sipp –sn uac 127.0.0.1 and ./sipp –sn uac 139.54.102.135 Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 2-60232(a)139.54.102.135) Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist The user sipp and service does'nt exist Added user: kamctl add sipp testpasswd kamctl add service testpasswd Still same error: Aug 27 10:52:17 localhost /usr/sbin/kamailio[56289]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist Aug 27 10:52:17 localhost /usr/sbin/kamailio[56284]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 977-61357(a)139.54.102.135) Please advise. Thanks and Regards.

1 1

kamailio stop registering agents
by Gaurav Bmotra 27 Aug '19

27 Aug '19

hi i m using kamailio 5.1,,, on ubuntu 18.4 installed form source .... after few days it stop registering client to kamailio.... it resolved after restart service ... same after few day loges ---- --------------------- kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Wed 2019-08-21 08:07:44 UTC; 5 days ago Docs: man:systemd-sysv-generator(8) Process: 1936 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [main.c:2636]: main(): processes (at least): 27 - shm size: 67108864 - pkg size: 838 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: dispatcher [dispatcher.c:813]: ds_warn_fixup(): failover functions used, but required AVP Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: <core> [core/tcp_main.c:2850]: tcp_init(): bind(c, 0x7fbabac2c1f4, 16) on 172.31.35.49:509 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 21 08:07:44 talk kamailio[1936]: * already running Aug 21 08:07:44 talk kamailio[1936]: ...done. Aug 21 08:07:44 talk systemd[1]: Started LSB: Start the Kamailio SIP proxy server. plz help thank you -- *Regards:* Gaurav Kumar

3 3

Re: [SR-Users] [sr-dev] Failed to catch some reply message with reply_route block
by Henning Westerholt 27 Aug '19

27 Aug '19

(moved the discussion to user list) Hello Salah, I see the this output in the log message: 6(26) INFO: <script>: on_reply_route [SIP/2.0 200 OK Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bKe68a.99dc492fe74dcfe1017828521a5fa362.0 2(22) INFO: <script>: on_reply_route [SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bK178a.79cddb135aabfc42736fd9e9fd826ba8.0 and so on.. So you are seeing the replies correctly, or I wrong? Cheers, Henning Am 22.08.19 um 21:29 schrieb Salah Ahmed: Hello Henning, We already added xlog in the onreply_route and found all messages in between kamailio and UAS. The attached log files in first email has those logs. If any other specific log needed I am here to provide them. Thanks, Salah On Thu, Aug 22, 2019 at 2:24 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, ok, see if you are able to output some easy log with "xlog" in this route, to make sure there is no cfg problem somewhere. Cheers, Henning Am 22.08.19 um 21:15 schrieb Salah Ahmed: Hello Henning, Thanks for quick reply, We armed the t_on_reply just before t_relay() in the route block. t_on_reply("LOGRPL"); if (!t_relay()) { sl_reply_error(); } Thanks, Salah On Thu, Aug 22, 2019 at 2:03 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, the replies are going from the UAS to the UAC over Kamailo as a proxy, with the exception of the hop-by-hop 100. So you should see the 200 OK in the Kamailio in reply_route and onreply_route. Maybe you can check if you armed the onreply_route with t_on_reply for the INVITE routing. Cheers, Henning Am 22.08.19 um 19:50 schrieb Salah Ahmed: Hello, We facing an issue on capturing sip message in Kamailio(Version: 5.2.3). The scenario is very simple. UAC Kamailio UAS |--------INVITE--------->| | | |-------INVITE------->| | |<------100 Trying----| |<-------100 Trying------| | | |<------200 Ok--------| |<-------200 Ok----------| | In this simple scenario, we can't catch any reply messages in between the Kamailio and UAC. We have tried reply_route, onreply_route, and onsend_route. But no one work to grep that reply on that side. onsend_route was bad try as its only for forwarded reply message. Is there any other magic to capture those replies. A debug=3 log message attached here. Thanks, Salah _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

4 11

exit stop procesing or not after a conditional? based on "exit" core docs seems does not explain behaviour
by PICCORO McKAY Lenz 26 Aug '19

26 Aug '19

in my kamailio-asterisk integration i have a exit after the authentication logic, BUT; there's another conditional after the "exit" sentence, so my ask, if there's a exit in that piece of code .. that next conditional will not be executed? Docs seems said "stop execution of script" so all the rest of the logic in file will not be routed? The documentation seems does not explain in good behavior what happened or the kamailio-asterisk wiki page does not have good explanations for that! it's that an error? https://www.kamailio.org/wiki/cookbooks/4.3.x/core#exit - - ``` if (from_uri==myself) {#!ifdef WITH_ASTERISK if (!proxy_authorize("$fd", "sipusers")) {#!else if (!proxy_authorize("$fd", "subscriber")) {#!endif proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } ``` 8:52 am Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

reply cannot be forwarded - no 2nd via
by Ryan McEvoy 25 Aug '19

25 Aug '19

Hello, I have been having an issue using kamailio to load balance to a SIP/MRCP server. I believe that the issue actually lies with the reply being sent from the final recipient, but I wanted to be sure that I understood the problem and to see if there were any possible solutions within Kamailio. I am using the dispatcher module to dispatch SIP INVITES to media resources, in this case I am just using a single host in a single dispatcher group. The configuration that I have has worked just fine for other SIP/MRCP servers, which is why I think the issue is with this target in particular. The INVITE is routed correctly and contains the headers that it should. The invite comes originally from a Freeswitch, arrives at Kamailio, who should then dispatch to a host in the group. x.x.x.x:4901 is Kamailio x.x.x.x:8090 is Freeswitch y.y.y.y:8000 is the final target This is the invite, as received by the final target after dispatched via Kamailio INVITE sip: x.x.x.x:4901 SIP/2.0 Record-Route: <sip: x.x.x.x:4901;transport=tcp;lr> Via: SIP/2.0/TCP x.x.x.x:4901;branch=z9hG4bK3a9a.7f9a4b588cd75108cfea9a2bd9ffa829.0;i=3 Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S Max-Forwards: 69 From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ To: <sip: x.x.x.x:4901> Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE User-Agent: tts Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Disposition: session Content-Length: 281 v=0 o=IPSoft 3367790083977298880 2789991278648035021 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:existing a=resource:speechsynth a=cmid:1 m=audio 32038 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=recvonly a=mid:1 The reply that the server sends back appears to be missing a second Via header, which is preventing Kamailio for routing it correctly (based on the error message from core/forward.c) This is the reply sent by the target destination SIP/2.0 200 OK Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S To: <sip: x.x.x.x:4901>;tag=ndszib3k From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE Contact: "Voiceware MRCP Server" <sip:vwmrcp@ y.y.y.y:8000;transport=tcp> Content-Type: application/sdp Content-Length: 324 v=0 o=- 1566492047 1566492047 IN IP4 y.y.y.y s=Voiceware MRCP Server 2.15.0.7. Dec 28 2018 c=IN IP4 y.y.y.y t=0 0 m=application 8002 TCP/MRCPv2 1 a=channel:12951125463@speechsynth a=connection:new a=setup:passive a=cmid:1 m=audio 30158 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=sendonly a=ptime:20 a=mid:1 It is correct that there should still be a 2nd Via header in the 200 OK reply from the target host, right? If so, is there anything else that can be done in Kamailio to handle this, or is this mrcp server simply not complying properly with SIP? There’s no funky natting between these hosts or anything like that. Thanks in advance!

2 1

Wrong ACL to Provider kamailio+asterisk+uac
by Zhan Bazarov 25 Aug '19

25 Aug '19

Hello! Please, is it possible to change from/to headers in ACK message to provider? this is exactly our case, please, look it when u get a chance. http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html…

2 13

set_rtpengine_set question
by Juha Heinanen 24 Aug '19

24 Aug '19

When set_rtpengine_set() is called with two parameters before rtpengine_offer(), e.g. set_rtpengine_set("1", "2"); rtpengine_offer(); it was not clear to me from rtpengine README, if the sets need to reversed when set_rtpengine_set() is called before rtpengine_answer(), i.e., set_rtpengine_set("2", "1"); rtpengine_answer(); or can the order of the sets be the same as it was in case of rtpengine_offer()? -- Juha

1 1

Dealing with optional SRTP
by Mark Boyce 24 Aug '19

24 Aug '19

Hi all (Using Kamailio + RTPEngine, both current stable versions) Just wondering what current best practice is for dealing with “optional SRTP” such as those generated by Yealink devices. (I know this breaks the RFC but lots of things seem to use this method.) Where the device SDP message looks like; m=audio 50004 RTP/AVP 8 97 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhJOWOXPuzMeBmrbx2wAKohuveAvZeMEJTzfE7GT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:yF+EhR3RtxSDCGtLCnEAl3Zrdut4b8cfFQP+hFqf An unencrypted RTP/AVP media line but with crypto lines showing that the device will support SRTP encryption but it is not mandatory. The system of offer SRTP and try and catch on not supported/acceptable here fail feels a little clumsy and relies on how gracefully the device sends “not acceptable here”. I’m wondering if I just short cut it and say that if a device offers a acceptable crypto line then Kamailio sends back a SAVP response. Naturally this question is then closely followed by how to do optional SRTP invites originating from Kamailio without just directly manipulating lines? Feels like there should be an RTPEngine setting for Optional? RTP/AVP+SAVP ? Thoughts? Mark

1 0

Kamailio as an Outbound Proxy - Strange Sip Flow
by Diogo Hartmann 23 Aug '19

23 Aug '19

Hello Kamailio experts, We have a Kamailio as outbound proxy in front of Asterisks PjSIP. There is a strange behavior after 200 OK: - ACK comes from Asterisk Public IP instead of coming from the internal IP address. - BYE can not be processed (404 not here) SNGREP print attached. Has anyone ever experienced this? I can provide more information if needed, just let me know what info. Thanks! Diogo Hartmann

2 5

Cant finish call
by przeqpiciel 23 Aug '19

23 Aug '19

Hi all, I am totally new with Kamailio but i have needs to build small cluster of Asterisk. Right now I could succesfully register to asterisk via kamailio and even make a call to hear tt-weasels but Asterisk cant close call My infrastructure diagram is here http://www.asciidraw.com/#Draw6816041626716434245/903175935 This is what probably I need is to know how to tell to Asterisk that send BYE to private IP of Kamailio instead public IP of Kamailio, am I right ?

1 0

TCP: Strange Device Behaviour
by Carsten Bock 23 Aug '19

23 Aug '19

Hi, I have a device here, which does some sort of strange things: I have SIP Message, e.g. with a length of 1100 Bytes and the devices adds "0" to the end to meet the MTU of the network (for a single TCP frame), some sort of TCP padding. The following TCP Frame contains only the next message. Looking at the trace in Wireshark, sngrep or similar, all of them try to re-assemble the next message in the TCP stream, but however don't decode them as SIP due to to first couple of zero's in the message. Looking at Kamailio, it seems not to have any issues with this and decodes/parses the message properly (they show up on Homer properly); however it also forwards the message INCLUDING the first couple of zero's even on UDP. To make it complitely irritating, the phone itself would ignore such packets, if we would forward them to the phone with starting zeros. Has anyone seen anything like this and is it standards-compliant? Any proper and easy way to get rid of the starting "zero's" from a message before forwarding these messages? I could always "hack" something into src/core/tcp_read.c, but I wonder if there is already a proper way? Any ideas anyone? Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

About STIR/SHAKEN - Caller Identity
by Daniel-Constantin Mierla 23 Aug '19

23 Aug '19

Hello, at couple of events I participated during the past few months, I was asked about support of STIR/SHAKEN (caller identity authentication/verification), which is a hot topic these days at least in USA, aiming to combat "fraudulent" robo-calling. Therefore I thought of share some details with everyone in the community about the state in Kamailio, writing to both devs and users, the information being relevant for everyone. We already have the (related) module named auth_identity, available since 2008 (iirc): - https://www.kamailio.org/docs/modules/stable/modules/auth_identity.html But it implements the previous iteration of the specs for caller identity, respectively RFC 4474: - https://tools.ietf.org/html/rfc4474 However, that RFC is obsoleted by 8224 (the latest core specs for STIR/SHAKEN): - https://tools.ietf.org/html/rfc8224 Then, there are also RFCs 8225 and 8226 to add to the core specs. Should anyone be interested to implement STIR/SHAKEN specs in a modules, I would suggest to start from auth_identity -- might not be much work to update it to become conform with latest specs (a new module can be created, of course, even when starting from auth_identity). However, these specs are about signing the SIP request (the INVITE) with special PKI certificate. It can be done easily with embedded scripts such as Lua or Python (inline execution in native kamailio.cfg or using kemi scripts). At Kamailio World 2019, one of the participants I discussed with told me they already implemented using Lua. That's it for a starting point, if anyone wants to discuss more, just reply to sr-users and add your comments or ask the questions. If someone wants to go ahead and work on a C module, announce yourself to avoid duplicate work of others, and use sr-dev if you need assistance on module development. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Unable to start kamailio_pcscf.service
by Jack R 23 Aug '19

23 Aug '19

1. Followed steps from Step # 3 to Step # 16 : https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… 2. When I run last command >systemctl start kamailio_icscf kamailio_pcscf kamailio_scscfthe root@ubuntu18-04:/etc/init.d# *systemctl start kamailio_icscf kamailio_pcscf kamailio_scscf* Job for kamailio_pcscf.service failed because the control process exited with error code. See "systemctl status kamailio_pcscf.service" and "journalctl -xe" for details. 3. All services start but not kamailio_pcscf: root@ubuntu18-04:/etc/init.d# *systemctl status kamailio_pcscf.service* ● kamailio_pcscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_pcscf; generated) Active: failed (Result: exit-code) since Mon 2019-08-19 09:05:30 UTC; 45min ago Docs: man:systemd-sysv-generator(8) Process: 52271 ExecStart=/etc/init.d/kamailio_pcscf start (code=exited, status=1/FAILURE) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: ERROR: bad config file (1 errors) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: * Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Control process exited, code=exited status=1 Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Failed with result 'exit-code'. Aug 19 09:05:30 ubuntu18-04 systemd[1]: Failed to start LSB: Start the Kamailio SIP proxy server. lines 1-16/16 (END) Please advise. Thanks and Regards.

2 10

basic Kamailio frontend to Asterisk
by Travis Ryan 22 Aug '19

22 Aug '19

All, I would like to use Kamailio as a frontend/proxy for an asterisk machine/cluster. I'm very well versed in Asterisk, but VERY noob to Kamailio. I've watched so many Kamailio presentations and feel like I've scoured the web for any article that could help me. There is alot out there, but I'm not finding my exact situation. I believe I understand the dispatcher stuff that I'll need down the road, but I'm having issues understanding the setup of my SIP trunk and Kamailio. I used UAC to setup the trunk connection in Kamailio and it sort of worked. I'm not sure what I do for Asterisk. I'd suspect that in this situation Kamailio would handle the registration and trunk connection for me. What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need to still register the trunk from each Asterisk box "thru" the Kamailio proxy, etc? Also, I'm merely accepting outside calls and then validating the caller and bridging them back out to the PSTN, so I don't have any local SIP clients, etc., so no need to register the sip devices, etc. Thanks! Travis

4 5

Docker image for 5.2.4
by Colin, Andrew 22 Aug '19

22 Aug '19

Hey All Who updates the images on https://hub.docker.com/r/kamailio/kamailio/tags i dont see the latest image kamailio:5.2.4-stretch --

1 0

{param.count,delimiter} issue
by Juha Heinanen 20 Aug '19

20 Aug '19

Looks like there is bug in {param.count,delimiter} transformation: $var(test) = "a=1;b=2"; xinfo("param count = $(var(test){param.count,;})\n"); $var(test) = "a=1,b=2"; xinfo("param count = $(var(test){param.count,,})\n"); syslog: Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 2 Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 1 -- Juha

3 10

Slowly increasing PKG memory usage
by Andrew White 20 Aug '19

20 Aug '19

Hi all, I had a Kamailio crash the other day, and some debugging showed I ran out of PKG memory. Since then I’ve run a simple bash script to compile the amount of memory used by all child processes, effective /usr/local/sbin/kamcmd pkg.stats | grep real_used summed together. I’ve graphed out the data, and there’s a clear growth of PKG memory going on, mostly increasing during our busier daytime hours. https://i.imgur.com/UTzx2k1.png Based on this, I suspect either a module loaded or something within my app_ruby conf is leaking memory. I’ve been reading through https://www.kamailio.org/wiki/tutorials/troubleshooting/memory <https://www.kamailio.org/wiki/tutorials/troubleshooting/memory>, but I’m a bit nervous, as I’m not really a C/deep memory type of guy. I can see a GDB script I can attach to Kamailio, but is that going to use significant resources to run or impact the running process? Is there a newer/better/alternative way to do this, and to help me break this down? Thanks! Andrew

3 17

Evapi dispatcher is consuming all the private memory
by José Seabra 20 Aug '19

20 Aug '19

Hello there, I'm using kamailio 5.2.1 with evapi module to send and receive events from an external application. During my stress tests to kamailio i have been noticing that at some point kamailio is getting out of memory: edge-sip-proxy[20565]ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fb8730b6010, 64); Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fb8730b6010, 64) called from core: core/parser/msg_parser.c: parse_headers(325), module: core; Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/parser/msg_parser.c:328]: parse_headers(): memory allocation error edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: pv [pv_core.c:704]: pv_get_callid(): cannot parse Call-Id header edge-sip-proxy[20565]: INFO: ESP_LOG: <null>: [evapi:connection-received] EVAPI msg received -> {"aor":"sip:FUZECI1-x1516@uc.fuze.domain ","contact":"<sip:FUZECI1-x1516@10.225.121.206:5060;transport=UDP>"} edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fb8730b6010, 64); Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fb8730b6010, 64) called from core: core/parser/msg_parser.c: parse_headers(325), module: core; Free fragment not found! The process id 20565 is running out of memory as you can see bellow: output of kamcmd pkg.stats entry: 14 pid: rank: -2 used: 6058696 free: 824 real_used: 8387784 total_size: 8388608 total_frags: 30 This process ID is related with Evapi Dispatcher kamcmd ps | grep 20565 20565 EvAPI Dispatcher After stop the sipp that was generating traffic to kamailio the evapi process still have all the memory consumed Can it be some memory leak in Evapi module? Best Regards José Seabra -- Cumprimentos José Seabra

2 12

HSS download - open source
by Jack R 20 Aug '19

20 Aug '19

Please could you advise where can I get open source HSS that works with latest Kamailio 5.2 release for OS: ubuntu18-04:~$ lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 18.04.3 LTS Release: 18.04 Codename: bionic If possible please share the steps to integrate with Kamailio. The link in this article for FHoSS repository is not working: https://www.kamailio.org/wiki/tutorials/ims/installation-howto If not Open source then maybe an evaluation software. Please advise. Thanks and Regards.

2 1

kamailio doesn't reply on internal interface
by Zhan Bazarov 19 Aug '19

19 Aug '19

Hello, we have interesting issue... Some time kamailio comes unreachable on private interface for asterisk, but port checked via telnet and everything is ok... ps. connection via public ip works good root@kamailio:/usr/local/etc/kamailio# lsof -i :5068 | grep LISTEN kamailio 6014 root 13u IPv4 8341589 0t0 TCP kamailio.local.ip:5068 (LISTEN) kamailio 6014 root 14u IPv4 8341590 0t0 TCP kamailio.public.ip:5068 (LISTEN) root@kamailio:/usr/local/etc/kamailio# telnet from asterisk machine: root@asterisk:/etc/asterisk# telnet kamailio.local.ip 5068 Trying kamailio.local.ip... Connected to kamailio.local.ip. Escape character is '^]'. But peer is unreachable in asterisk... More than, kamailio comes reachable when I change transport from TCP to UDP, or when I restart kamailio daemon(but it situation repeats every day).. Any ideas how I can fix it? thanks!

2 1

set_rtpengine_set(set1, set2) issue
by Juha Heinanen 19 Aug '19

19 Aug '19

My sip proxy has two rtpengine sets: # sip-proxy_ctl rtpengine.show all { url: udp:192.26.134.1:6050 set: 0 index: 0 weight: 1 disabled: 0 recheck_ticks: 0 } { url: udp:192.26.134.40:6050 set: 1 index: 1 weight: 1 disabled: 0 recheck_ticks: 0 } Config makes callBefore calling rtpengine_offer(), config calls set_rtpengine_set("1", "0"); rtpengine_offer("..."); As result, debug shows that sip proxy sends TWO offer commands to the rtpproxy in set 1 and NONE to the rtpproxy in set 0. Debug at set 1 engine: Aug 19 18:06:08 buster-10-0 rtpengine[1314]: NOTICE: [5e1407211cc50fd9]: Creating new call Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Replying to 'offer' from 192.26.134.1:33284 (elapsed time 0.000812 sec) Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Received command 'offer' from 192.26.134.1:33284 Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Replying to 'offer' from 192.26.134.1:33284 (elapsed time 0.005247 sec) I haven't been able to figure out, why that is the case. It is a bug, or am I missing something? I added INFO call to set_rtpengine_set_f(): INFO("set1/set2 = %u/%u\n", selected_rtpp_set_1->id_set, selected_rtpp_set_2->id_set); return 1; and it correctly printed: Aug 19 18:06:08 salmon /usr/bin/sip-proxy[26911]: INFO: rtpengine [rtpengine.c:3574]: set_rtpengine_set_f(): set1/set2 = 1/0 Any ideas? -- Juha

1 0

MOH setup
by Maryam Baghdadi 19 Aug '19

19 Aug '19

Hello everyone, I need to setup Music on hold feature and I did all modifications in configuration file and did all steps for setup: 1- load MOH and all requirement modules . 2- rtpproxy setups and works correctly 3- mohqueue and mohqcalls tables created. 4- audio file with type index (novanet.8) exists 5- parameters of MOH module setup and mohq_process() function called in request_route()‌. 6- table mohqueue initiated like this‌: +----+------+----------------------------+-------------------+---------+-------+ | id | name | uri | mohdir | mohfile | debug | +----+------+----------------------------+-------------------+---------+-------+ | 1 | main | sip:main at 191.38.245.2 | /var/kamailio/MOH | novanet | 1 | +----+------+----------------------------+-------------------+---------+-------+ Kamailio starts successfully and no error appears , call setup but on hold doesn't work and no audio file plays. I review all previous mailing lists responses but no one answers completely and before me others had this problem.

4 5

Replacing Caller ID
by Pete Kay 18 Aug '19

18 Aug '19

Hi Could anyone tell me if there is a module for replacing the caller ID in the SIP messages? I need to replace the caller ID when the call comes in and leaves Kamailio. When the egress sends back any response, I need to revert the caller ID back to its normal form. I have a long list of numbers to be used as caller-ID, and I need to keep modifying them without having to restart Kamailio. Do I need to write my own modules for maintaining the list of numbers to be used as caller-id? I am running at 2000 CPS, so I think I need to implement some sort of hash table in order for the caller id to be retrieved in round roubin fashion very quickly. Is there any modules for this purpose or I need to write a cusom one? Pete

2 1

Re: [SR-Users] Wildcarded IMPU
by Henning Westerholt 17 Aug '19

17 Aug '19

Hello Maryam, (please keep the list in CC) understand. There is work ongoing to update the IMS modules (e.g. refer to https://www.kamailio.org/wiki/features/new-in-devel). I don't know about a current effort to add wildcarded AVPs to them, though. If you like to implement something in this regards, it is always good to announce it quickly on the developer list sr-dev to get some feedback and to to synchronize with other developers. Cheers, Henning Am 17.08.19 um 08:04 schrieb Maryam Baghdadi: Hello Henning, Thank you for your response . I read all ETSI‌ standards , latest version of standards are according to Release15 IMS but IMS modules in kamailio support Release 7 IMS. for example there is Wildcarded_IMPU AVP in AUR in R15 IMS which isn't implemented in IMS_icscf module in kamailio . I know all AVP lists are exist in kamailio and I can add my requirements to modules code . I just wanted to know ims modules in kamailio will be updated or not? On Wed, Aug 14, 2019 at 11:55 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Maryam, Kamailio supports AVPs and also an extended version called XAVP. About your question - I quickly looked into ETSI TS 129 329 and TS 129 229 but they did not define how this actually works.. Can you elaborate what you actually are looking for? Cheers, Henning Am 05.08.19 um 12:41 schrieb Maryam Baghdadi: hello, I need to know wildcarde_IMPU feature support in kamailio or not? I use kamailio as scscf and need to have support of wildcarded_impu in my scscf. is there any implementation of AVP_Wildcarded_IMPU in kamailio ? thanks _______________________________________________ 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 -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

uac_replace_from() and multiple branches
by John Quick 16 Aug '19

16 Aug '19

v5.2.3 of Kamailio. Serial forking is used to route calls to a multiple destinations as described in documentation for the TM module. Also using the DIALOG module - my script calls dlg_manage() before loading the serialised branches. I need to be able to substitute the username part of the From header and have been trying this with calls to uac_replace_from(). modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) There is no problem if all branches use one substituted From username, in which case my script calls uac_replace_from() in the main route. uac_replace_from("sip:" + $avp(route:src) + "@" + $fd); However, I may need to substitute a different value for each branch. Can this be done? I tried moving the call to uac_replace_from() into the branch route, but then I get an error when the final BYE is received: ERROR: uac [replace.c:281]: replace_uri(): decline FROM replacing in sequential request in auto mode (has TO tag) Is there a simple solution to this please? Hoping I can still use UAC auto restore mode because manual restore looks quite tedious/complex. Thanks. John Quick Smartvox Limited www.smartvox.co.uk

3 3

pike parameters doup when have dinamyc ip clients and scanners
by PICCORO McKAY Lenz 15 Aug '19

15 Aug '19

I read documentation for pike usage.. and have a doub.. what it's the best for very dinamyc ip of my devices.. i mean, i'm just curious of that very ironic and problematic scenario: for the scanners i setup fail2ban but only when the scaning are detecte.. but if i have the pike option how this: # this it's my setup for pike due the dinamyc ip and devices over the internet: modparam("pike", "sampling_time_unit", 4) modparam("pike", "reqs_density_per_unit", 80) modparam("pike", "remove_latency", 60) ... route { if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike block $rm from $fu (IP:$si:$sp)\n"); exit; } ... } I put the remove latency in 60, so then due are dinamycally must remian in memory more (due any one will be a possible clilent), just ban if there are 180 (60*3) request each 4 seconds, it's a good configuration or maybe i'm wrong please help me! -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 6

RFC 3860 support - Kamailio is sending SUBSCRIBE back with extra Via header
by Praveen Vatti 15 Aug '19

15 Aug '19

Hi Kamailio support, Basically, I am looking for RFC3860 support - i.e., accept SUBSCRIBE for reg event and send 200OK, NOTIFY to client. With some browsing on Internet I came across Kamailio, which is supporting RFC3860. I have installed and attached is my CFG file. I am sending a SUBSCRIBE request with reg event to Kamailio and it is sending back the same request with extra Via header. Via now contains Kamailio system IP and the source of SUBSCRIBE IP. Please help me in resolving this issue. I have loaded db_mysql, dialplan, tm, sl, presence, presence_reginfo modules. Thanks, Praveen

2 6

Wildcarded IMPU
by Maryam Baghdadi 14 Aug '19

14 Aug '19

hello, I need to know wildcarde_IMPU feature support in kamailio or not? I use kamailio as scscf and need to have support of wildcarded_impu in my scscf. is there any implementation of AVP_Wildcarded_IMPU in kamailio ? thanks

2 1

Uac_replace for header's parameters
by Duarte Rocha 14 Aug '19

14 Aug '19

Greetings, I'm using uac_replace_to() and uac_replace_from() to do some number manipulations. It allows me to modify and automatically restore Display Number and URI. This works nicely, however it only allows to manipulate URI's parameter. Is there any method to the same and include the Header's parameters too? Best regards, Duarte Rocha

2 1

Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation
by Joel Serrano 14 Aug '19

14 Aug '19

Hello, Simple doubt regarding dispatcher module when you have the ds_default_socket() modparam set: Say you have 2 kamailio boxes, active + passive, one shared IP via keepalived.... On the actrive node, you have dispatcher.send_ping to 1, and the passive has it set to 0. So far OK, only the active box is sending out probing OPTIONS requests, and it's using as outbound socket the virtual IP out of all the available ones (as defined with ds_default_socket() modparam) I have seen in a capture that the outbound OPTIONS look like: OPTIONS sip:190.14.203.219:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.180 ;branch=z9hG4bK7abb.6fbaccc1000000000000000000000000.0 To: <sip:190.14.203.219:5060> From: <sip:dspinger@my.domain>;tag=e2bdd495733c59fd91487a137fccad4e-8a73 CSeq: 10 OPTIONS Call-ID: 2019f8491329c382-31419(a)A.B.C.181 Max-Forwards: 70 Content-Length: 0 A.B.C.180 -> VRRP A.B.C.181 -> Physical IP of the box My doubt is, shouldn't ds_default_socket also update the IP used to generate the Call-ID used in the OPTIONS request? Currently, I believe it's using the first defined listen= IP from config script as I switched the order the listen= are defined and I get the desired result: OPTIONS sip:186.188.220.174:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.180 ;branch=z9hG4bKc97e.a8d9e1c2000000000000000000000000.0 To: <sip:186.188.220.174:5060> From: <sip:dspinger@my.domain>;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323 CSeq: 10 OPTIONS Call-ID: 7d9a92c218fc1ba0-32111(a)A.B.C.180 Max-Forwards: 70 Content-Length: 0 Is this expected? Cheers, Joel.

3 4

Kamailio/Freeswitch auth user and UserID
by Petarr Jones 14 Aug '19

14 Aug '19

Hi I am new to Kamailio and would like some guidance on linking up with Freeswitch. I have an exiting hosted Asterisk PBX that uses a combination of extension and pbx username (eg 1001-126252581276) and I want to move it over to Kamailio/Freeswitch without re-configuring the SIP phones. I have tried Freeswitch "number-alias" but it does not seem like a good fit due to things like BLF not working and the added overhead etc. Is there a way to configure Kamailio to authenticate a user based on the above username and then send an extension to Freeswitch as the UserID instead, thereby separating the Kamailio authentication from the Freeswitch UserID? As I don't yet have a full understanding of the Kamailio/Freeswitch interaction, I don't know whether this will cause other issues? Thanks in advance Petarr

2 8

Re: [SR-Users] Dispatcher round-robin problems in 5.2
by JR Richardson 14 Aug '19

14 Aug '19

Hey Alex, After I recently upgraded, I had to include more response codes, 5.2 by default only 200 OK is accepted. I had some upstream carriers sending 405 and were taken off-line as a result until I added that code in modparm: modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=405") Not sure if this applies to what you are doing, but my dispatcher round robin distribution acts as it did prior to upgrade to 5.2, pretty even, no anomalies with any nodes getting more calls than any other. JR JR Richardson Engineering for the Masses Chasing the Azeotrope Hi, I've got a proxy that does very basic round-robin distribution of calls to an elastic pool of servers, defined in a simple SQLite database with equal priorities. Nothing too complicated, just a vanilla `ds_select_domain("1", "4")`. The only complication is that seldom are most of the gateways enabled. Probing stuff was turned on, but didn't do quite what was desired, so it was replaced with a system that manages activation and deactivation of the gateways externally. Nevertheless, these are the left-over probing settings: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:pinger@domain") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshold", 256) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 0) After an upgrade from 5.1 to 5.2.3 (009f62), the round-robin has become extremely "sticky" and tendentious. That is to say, it directs _almost_ all calls to a single gateway -- the first gateway in the table -- though not 100%. There is also a marked tendency to send lots of consecutive calls to the same gateway, even if it's not the first one, though most of the time it is. Has anyone experienced similar upon upgrading to 5.2? Thanks much! -- Alex

2 2

Wrong Record-Route and Via header fields when TCP is used
by Leonid Fainshtein 14 Aug '19

14 Aug '19

Hi, Kamailio server has two legs that are connected to different networks. I'm using Kamailio v.5.2.3 and the "enable_double_rr" is implicitly set to "1". The leg "A" IP address is 10.159.65.1 The leg "B" IP address is 192.168.0.31 The call is initiated from 10.159.65.18 According to the "rr" module documentation, function record_route() should insert two "Record_Route" header fields when a request is accepted on one leg is should go out via the second leg. This works as expected in case of UDP protocol: INVITE sip:2000@192.168.6.106:5460;transport=UDP SIP/2.0 Record-Route: <sip:192.168.0.31;r2=on;lr;did=e2c.a191> Record-Route: <sip:10.159.65.1;r2=on;lr;did=e2c.a191> Via: SIP/2.0/UDP 192.168.0.31;branch=z9hG4bKcfa5.d64ecbd87d5315b5993c4ccf16f86537.0 Via: SIP/2.0/UDP 10.159.65.18:5060;rport=5060;branch=z9hG4bK3a9e9a4d But when the TCP protocol is used then the outbound message looks like this: INVITE sip:2005@192.168.0.178:35058;transport=tcp SIP/2.0 Record-Route: <sip:10.159.65.1;transport=tcp;lr;did=bb6.7dc1> Via: SIP/2.0/TCP 10.159.65.1;branch=z9hG4bKc85a.14afc3867dd3220826f9b9940f78168f.0;i=3 Via: SIP/2.0/TCP 10.159.65.18:5060;rport=58616;branch=z9hG4bK1469331f There are two problems there: a) only one Record-Route with leg is inserted b) the added "Via" header field contains the leg "A" IP address instead of expected leg "B" IP address (192.168.0.31). In the LAN trace I see that in reality the message was sent from leg "B". Is it a bug? Best regards, Leonid Fainshtein

2 13

Kamailio v5.2.4 Released
by Daniel-Constantin Mierla 14 Aug '19

14 Aug '19

Hello, Kamailio SIP Server v5.2.4 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.4. For more details about version 5.2.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/08/kamailio-v5-2-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Planning the release of Kamailio v5.2.4
by Daniel-Constantin Mierla 14 Aug '19

14 Aug '19

Hello, I am considering to do soon a new release out of branch 5.2, respectively v5.2.4. I am aiming for August 12 or August 14, a matter of how backporting is finished, given that next week some devs travel to ClueCon. As usual, if you are aware of issues not reported yet on bug tracker or fixes not yet backported to branch 5.2, act during the next days to have enough time to approach them. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 3

Strange behavior for unregister() function. Wrong docs? Bug?
by Joel Serrano 13 Aug '19

13 Aug '19

Hello, I'm playing around with the unregister() function from the registrar module: https://www.kamailio.org/docs/modules/devel/modules/registrar.html#registra… Docs say that that it can be used in REQUEST_ROUTE or FAILURE_ROUTE. I needed it in REPLY_ROUTE, and before reading the docs I tried it and got this error: Aug 12 15:52:44 csbc01 kamailio: CRITICAL: <core> [core/cfg.y:3526]: yyerror_at(): parse error in config file /etc/kamailio/csbc/reply_routes.cfg, line 27, column 71: Command cannot be used in the block#012 Aug 12 15:52:44 csbc01 kamailio[12737]: ERROR: bad config file (1 errors) But, I have this working on another setup, I compared, and my difference is that I'm running unregister() inside a route, and calling that route from reply_route, and that works (and by works, I mean it also does what it is supposed to do, which is unregister the AoR). So this doesn't work: ... onreply_route[MANAGE_REG_REPLY] { if(status=~"2[0-9][0-9]") { if(($sel(contact.expires)==0) || ($hdr(Expires)==0)){ if (unregister("location", "$sht(sipserver=>$fU::contact)")) { xlog("L_NOTICE", "[usrloc] removed user location\n"); } route(DEL_SIPSERVER); } ... But this works: ... onreply_route[MANAGE_REG_REPLY] { if(status=~"2[0-9][0-9]") { if(($sel(contact.expires)==0) || ($hdr(Expires)==0)){ route(UNREG_CUSTOMER); route(DEL_SIPSERVER); } ... ... route[UNREG_CUSTOMER] { if (unregister("location", "$sht(sipserver=>$fU::contact)")) { xlog("L_NOTICE", "[usrloc] removed user location\n"); } } ... So I'm in no-mans-land right now, docs say it's not allowed in REPLY_ROUTE, and if I try it it will fail, but if I try it inside a route, it woks (hack?). What is the correct approach here? Should unregister() be valid in reply_route (like other similar functions, for example: save()) and therefor docs should be updated and the error in the logs should also be updated, or if it's really not supported, shouldn't Kamailio not let me use it anywhere that a ends up being called from reply_route? I hope I have explained myself correctly. As always, please let me know if this should be a GH issue instead and I'll happily create it. Thanks, Joel.

3 5

Dispatcher round-robin problems in 5.2
by Alex Balashov 12 Aug '19

12 Aug '19

Hi, I've got a proxy that does very basic round-robin distribution of calls to an elastic pool of servers, defined in a simple SQLite database with equal priorities. Nothing too complicated, just a vanilla `ds_select_domain("1", "4")`. The only complication is that seldom are most of the gateways enabled. Probing stuff was turned on, but didn't do quite what was desired, so it was replaced with a system that manages activation and deactivation of the gateways externally. Nevertheless, these are the left-over probing settings: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:pinger@domain") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshold", 256) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 0) After an upgrade from 5.1 to 5.2.3 (009f62), the round-robin has become extremely "sticky" and tendentious. That is to say, it directs _almost_ all calls to a single gateway -- the first gateway in the table -- though not 100%. There is also a marked tendency to send lots of consecutive calls to the same gateway, even if it's not the first one, though most of the time it is. Has anyone experienced similar upon upgrading to 5.2? Thanks much! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 4

Kamailio DRA - Unable to get diameter_request() function to send
by Nick Jones 10 Aug '19

10 Aug '19

Hi Kamailio users, I'm trying to setup Kamailio as a Diameter Routing Agent, but I just can't seem to get the diameter_request() function to actually send a diameter request, I just get a warning about no JSON Response: *WARNING: ims_diameter_server [avp_helper.c:341]: addAVPsfromJSON(): No JSON Response* Inside my event route I've got one line to send the data to the peer "ims-hss.localdomain": *event_route[diameter:request]{ diameter_request("ims-hss.localdomain", $diameter_application, $diameter_command, $diameter_request); xlog("Forwarded Diameter Request");}* Hard as I try I can't seem to get this request forwarded to this peer and see that next xlog() line. In the config file I've included the cdp_check_peer & cdp_has_app checks to confirm the peer is up, and able to accept the request, (full config in pastebin below) I've defined the peers in the diametercfg.xml config file, and they're all showing as online when I do a "kamcmd cdp.list_peers": * FQDN: ims-hss.localdomain Details: { State: I_Open Disabled: False Last used: 0 Applications: { appid:vendorid: 16777216:10415 appid:vendorid: 16777216:4491 appid:vendorid: 16777216:13019 appid:vendorid: 16777216:0 appid:vendorid: 16777217:10415 appid:vendorid: 16777221:10415 } }* The source of avp_helper.c shows the warning is called if the length of the JSON is <= 0, but as I'm feeding back out what I've received it's not 0, an no recent major changes to the source, so I'm stumped as to why it's catching this. I'm running Kamailio 5.1.2 on Ubuntu 18.04 installed from Repos. Full Kamailio Config: https://pastebin.com/afgqUfWr Diameter CDP Config XML: https://pastebin.com/bVrBG8mG Relevant Syslog: https://pastebin.com/ZY8z2kd4 Kamcmd cdp.list_peers: https://pastebin.com/cKi4JAHC Any ideas as to what am I missing? Thanks in advance! Nick

1 0

need to Configure S-CSCF, I-CSCF and P-SCF
by ganesh nanaiah 10 Aug '19

10 Aug '19

Hi how do we configure the S-CSCF, I-CSCF and P-CSCF in kamalio 5.2.3 to authenicate with the HSS/PCRF, please point us to document to configure the IMS modules regards Ganesh

3 2

kamailio as UAC and NAPTR / sendsocket question
by Karsten Horsmann 09 Aug '19

09 Aug '19

Hi List, I try to register to Deutsche Telekom and there product Deutschland Lan siptrunk. Thats works find but i see an intressting behaviour on selecting the right outgoing interface. Kamailio is sending out with tcp the REQUEST via first private ip configured on that server (172.20.120.53). There is no listen directive for that. I forced NAPTR to use tcp or udp and i assume that kamailio got the right dns answers. On the list i read also that i can use force_send_socket to force the outgoing request. Now my idea - hey i use the $rP for the outgoing to select the right outgoing listen directive. $rP - reference to transport protocol of R-URI But to my surprise the logfile told me thats "UDP" - it sends out via TCP (thats okay). Whats an good transport selector variable from kamailio that works? event_route[tm:local-request] { if(!is_method("OPTIONS")) { xlog("L_INFO", "[tm:local-request] request [$rm] from [$fu] to [$ru] [$rP]\n"); } } INFO: <script>: [tm:local-request] request [REGISTER] from [ sip:+49XXXXXXXX@sip-trunk.telekom.de] to [sip:sip-trunk.telekom.de] [UDP] listen=tcp:2xx.xx.xx.xx:5060 listen=udp:2xx.xx.xx.xx:5060 listen=tls:2xx.xx.xx.xx:5061 advertise CFG_EXT_NAME:5061 listen=udp:172.20.120.55:5060 listen=udp:172.20.120.56:5060 listen=udp:172.20.120.57:5060 listen=udp:172.20.120.58:5060 listen=tcp:172.20.120.58:5060 use_dns_cache=on # use internal DNS cache use_dns_failover=on # depends on internal DNS cache dns_srv_loadbalancing=on dns_try_naptr=on dns_retr_time=1 # seconds before retrying a DNS request dns_retr_no=3 # number of DNS retransmissions dns_naptr_ignore_rfc=yes # ignore target NAPTR priority dns_tcp_pref=30 # TCP has second-highest priority dns_udp_pref=10 # use UDP with least priority tcp_connection_lifetime=3605 # set higher than registration expires #dont' restore modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) ## UAC REGISTER #!ifdef WITH_UAC_REGISTER modparam("uac", "reg_contact_addr", "CFG_PROD_IP") modparam("uac", "reg_timer_interval", 10) modparam("uac", "reg_retry_interval", 10) modparam("uac", "reg_db_url", DBURL) modparam("uac", "restore_mode", "none") modparam("uac", "auth_username_avp", "$avp(AVP_AUTH_USERNAME)") modparam("uac", "auth_password_avp", "$avp(AVP_AUTH_PASSWORD)") modparam("uac", "auth_realm_avp", "$avp(AVP_AUTH_REALM)") #!endif ip a l 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN group default qlen 1000 link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo valid_lft forever preferred_lft forever inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: ens192: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP group default qlen 1000 link/ether 00:50:56:b5:c1:48 brd ff:ff:ff:ff:ff:ff inet 172.20.120.53/24 brd 172.20.120.255 scope global ens192 valid_lft forever preferred_lft forever inet 2xx.xx.xx.xx/29 scope global ens192 valid_lft forever preferred_lft forever inet 172.20.120.56/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.57/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.58/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.55/24 brd 172.20.120.255 scope global secondary ens192 valid_lft forever preferred_lft forever inet6 fe80::250:56ff:feb5:c148/64 scope link valid_lft forever preferred_lft forever default via 172.20.120.253 dev ens192 172.20.120.0/24 dev ens192 proto kernel scope link src 172.20.120.53 2xx.xx.xx.xx/29 dev ens192 proto kernel scope link src 2xx.xx.xx.xx -- Kind Regards Mit freundlichen Grüßen *Karsten Horsmann*

4 9

failure_route - no UAC support
by Attila Czébán 09 Aug '19

09 Aug '19

Hello, I'm trying to restart phones with local-requests (from config or kamctl cisco_restart). I can enable t_on_failure("REMOTE_AUTH") in event_route[tm:local-request] but when 401 Reply comes in I get this warning: "tm [t_reply.c:968]: run_failure_handlers(): no UAC support (1, 0)". kamailio -V version: kamailio 5.1.8 (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 with gcc 6.3.0 How can I make it work? Thank you! Regards, Attila

1 0

Kamailio auth via http post.
by Zhan Bazarov 09 Aug '19

09 Aug '19

Hello, Its possible to made authentication via http request to third party Web-API instead of storing users credentials into Dataqbase? Now I looking on auth_ephemeral, but not sure is that my case... Don't see when I have to put API url.. Could anyone, please, shed any light on this issue?

2 1

XMLRPC request URL
by David Villasmil 08 Aug '19

08 Aug '19

Hello guys, I can't find any reference to a pseudovariable containing the request url, i.e.: http://myserver/XMLRPC?server=1.2.3.4 I'd like to get that server variable and manipulate the dispatcher list... the http client can't do XMLRPC or add a body, just an https request.. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

Breaking HTABLE changes from v5.2 to v5.3? Function renamed
by Joel Serrano 08 Aug '19

08 Aug '19

Hi all, Just noticed that when updating from 5.2 to 5.3 I saw some errors in Kam logfile regarding HTABLE module commands failing, the reason is the function 'sht_has_name' has been renamed to 'sht_match_name': https://www.kamailio.org/docs/modules/stable/modules/htable.html#htable.f.s… https://www.kamailio.org/docs/modules/devel/modules/htable.html#htable.f.sh… Not sure if updating this doc is automatic or manual but I think it's definitely worth mentioning: https://www.kamailio.org/wiki/install/upgrade/stable-to-devel Just FYI! Cheers, Joel.

2 1

Re: [SR-Users] Sqlops
by arish haque 08 Aug '19

08 Aug '19

Hi, Passing the value within single quotes works. Thanks for the help Fred. With Regards, Arish

1 0

Reload geoip2 database without restart
by Joel Serrano 07 Aug '19

07 Aug '19

Hi guys, Is there a way to instruct the geoip2 module to reload its database without a full restart of kamailio? Maybe some rpc command or something? I haven't found anything in the docs other than the comment about "Database is loaded at startup in cache." which makes me assume that just overwriting the db file wouldn't be enough. Any suggestions? Thanks!! Joel.

2 2

Sipt redirection number retrieves old values
by Duarte Rocha 07 Aug '19

07 Aug '19

Greetings, I'm doing VoiceMail detection using $sipt(redirection_number) from SIPT module and i've been having some false positives on the detection. I inspected further and noticed that in some cases $sipt(redirection_number) returns old values from previous messages, which makes me think that it's value is not being correctly refreshed. I'm using this variable on a on_reply_route. As far as i know, $sipt(redirection_number) is read only and can't be refreshed by me in the code. Here is an example : Aug 7 16:51:49 dev_sbc_a_vts01 /usr/local/kamailio-5.2/sbin/kamailio[4746]: ERROR: <script>: VMD2 : Reply Code : 180 ; Sipt Redir = 9610050000 and Call ID = 00158FED-98B7-E911-8B3B-4DA50AE2ED72(a)192.168.24.86 and Cseq = 83 Here is the JSON from the 180 message : https://pastebin.com/zFHKxyav As you can see, there isn't any 9610050000 in the message. What can be causing this issue ? Best Regards, Duarte Rocha

2 2

Sqlops
by arish haque 07 Aug '19

07 Aug '19

Hi all, I am querying mysql db inside routing block in kamailio.cfg ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Unknown column '8d77' in 'where clause' (1054) Aug 7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Aug 7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [select * from vca_asterisk where front_id=8d77] I'm getting the above error message when, variable front_id=8d77 is alphanumeric. It's working fine when front_id is pure numeric. eg: front_id=877 Database column front_id is VARCHAR(255). Any suggestions to solve this error? Thanks, Arish

2 1

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