- sr-users - kamailio.org

dispatch REGISTER over TLS
by sthustfo 10 Nov '19

10 Nov '19

I have a basic setup where kamailio receives SIP over websocket (no WSS) and forwards to SIP server over TLS. I have enabled TLS in kamailio.cfg and added dispatcher node as sips:SIP_SERVER:5061 and transport=tls. +----+-------+------------------------+-------+----------+---------------+----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+------------------------+-------+----------+---------------+----------------+ | 4 | 1 | sips:10.0.0.100:5061 | 0 | 0 | transport=tls | SIP SERVER | +----+-------+------------------------+-------+----------+---------------+----------------+ Now when REGISTER is received over websocket, kamailio is responding with error code 500 and phrase "500 I'm terribly sorry, server error occurred (7/SL)". And on the console I see the following error messages. 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: f1ecf7bcb659b07fe81e332e100044e5 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [ut.h:315]: uri2dst2(): no corresponding socket found for "10.0.0.100" af 2 (tls: 10.0.0.100:5061) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:467]: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -7 (-7) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:352]: t_relay_to(): -7 error reply generation delayed 12(33858) exec: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=587 a=24 n=sl_reply_error 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/msg_translator.c:162]: check_via_address(): (10.0.0.14, hsvmphm3ps12.invalid, 0) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} websocket [ws_conn.c:452]: wsconn_get(): wsconn_get for id [3] *tls.cfg contents* [client:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /home/test/kamailio/internal.key certificate = /home/test/kamailio/internal.crt ca_list = /home/test/kamailio/ca_list.pem Any reason why this error is seen? Any inputs appreciated. Thanks.

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Re: [SR-Users] _w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I added $du=$null; before route(RELAY) as you suggested but I get the same results: Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: transaction canceled Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) Regards, Konstantinos > On Fri Nov 8 10:24:06 CET 2019 Daniel Tryba wrote: > >> On Thu, Nov 07, 2019 at 04:53:43PM +0200, Konstantinos Merentitis wrote: >> >> I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: >> > You appear to do the same thing I do, except setting $du to null. If the user is REGISTERed $du will be set to their endpoint, changing $ru will have no effect on routing. So add $du=$null; in the TOVOICEMAIL route before route(RELAY) and see what happens. >

3 4

_w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I would like to forward all incoming calls that are not answered from kamailio, to my asterisk voicemail server. - I have not setup any Realtime Integration. I am using the "Redirected Dialed Number Information Service" header and calling "554(a)192.168.1.134" which is my voicemail server. Forwarding succeeds when the subscriber is not registered to kamailio: Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============User sip:6297@192.168.1.131 not found here, going to voicemail============= Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: transaction canceled Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) failure and voicemail routes are slightly modified from the default config: 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")) { xlog("=============User $ru not found here, going to voicemail=============\n"); $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; } 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")) { $var(RDNIS)=$di; xlog("\n RDNIS = $var(RDNIS) \n"); if($avp(oexten)==$null) return; # $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail1, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } else { if($rU==$null) return; # $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail2, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } route(RELAY); exit; #!endif return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { #modified here xlog("transaction canceled\n"); route(TOVOICEMAIL); #exit; #/modified } #!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; xlog("============= Route failure, forwarding to voicemail ============="); route(TOVOICEMAIL); exit; } #!endif }

2 1

How to read some data inside the INVITE ..
by Laura 07 Nov '19

07 Nov '19

Dears.. how is possibile to read that data 9052562462354464 inside the first INVITE message === Example === U 2019/11/07 14:46:59.320936 192.168.1.1:5060 -> 192.168.1.242:5060 INVITE sip:9052562462354464@192.168.1.242 SIP/2.0. Record-Route: <sip:192.168.1.1;lr;ftag=105854496;did=24e.e751;vst=AAAAAAAAAwQAAAAAAAAAAAAAAAAADQIAAwEDAAAHDwAGBAg1MDYw;nat=yes>. Max-Forwards: 66. Via: SIP/2.0/UDP 51.254.158.40:5060;branch=z9hG4bK501c.87a8e402c8f1cf93192ddd8e6744da57.1. From: "38617774738" <sip:38617774738@10.1.1.1>;tag=105854496. To: <sip:562462354464@192.168.1.242>. Call-ID: 1573138018-1602381967(a)10.1.1.182. CSeq: 20215490 INVITE ..... ETC ETC... Thanks to all

4 4

Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

3 5

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*

2 2

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; }

3 2

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

2 3

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!

1 0

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

4 3

$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*

3 2

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

4 4

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

3 8

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).

2 6

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

2 1

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.

3 8

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.

2 3

[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/

2 4

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

3 3

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.

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