sr-users December 2015

sr-users@lists.kamailio.org

70 participants
110 discussions

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v4.2.7 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.7 (or to 4.3.x series). For more details about version 4.2.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/12/kamailio-v4-2-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.4. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

8 years, 4 months

Kamailio as a SBC

by Guillaume

Hi everyone, I'm trying to make kamailio works as a SBC in front of Asterisk. I have read the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and it is close to what I'm looking for but still a little different. I want don't want kamailio to do auth and I want it to be a media gateway. The media and SIP path have to look like that: Phone 1 ----- kamailio -----Asterisk ---- Kamailio ---- Phone 2 First I have add an outboundproxy field in the Asterisk configuration to make all SIP messages from Asterisk passe through Kamailio. Then I started a minimal conf for Kamailio and it works at least for the SIP part. Call can be established perfectly. However the RTP proxy part is a mess. Kamailio changes the IP address in the sdp part only for INVITE messages and not for the 200 OK responses, so there is only one side of the media passing through Kamailio, the other side is sent directly to asterisk. But even for the half part rtpproxy craches with a segfault and Kamailio doesn't transmit this media. I have been searching some helps on google but nothing so far :/ I hope that someone here could help me. ----------------------------------------------- #!KAMAILIO ####### Defined Values ######### #!define ASTERISK 192.168.51.43 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=2 auto_aliases=no # to avoid simple attacks, use unconventional port port=5060 ####### Modules Section ######## mpath="/usr/local/lib/kamailio/modules" loadmodule "db_text.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "path.so" loadmodule "rtpproxy.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) modparam("path", "use_received", 1) modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## route { xlog("L-ALERT", "new message !!!!!!!!!!!!!!!"); if (!sanity_check()) { exit; } if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); xlog("L_NOTICE", "$si $rm $ru -- too many hops\n"); exit; } if (af==INET) { force_rport(); if (src_ip != ASTERISK) { $du = "sip:192.168.51.43:5060"; add_path_received(); route(RTPPROXY); record_route(); } else { loose_route(); route(RTPPROXY); record_route(); } } forward(); } route[RTPPROXY] { rtpproxy_manage("cwei"); } onreply_route { if (af==INET) { rtpproxy_manage("cwei"); } return(1); } --------------------------------------------- Guillaume

8 years, 4 months

Planning to release v4.2.7

by Daniel-Constantin Mierla

Hello, later this week, on Thursday, Dec 17, I am planning to release a minor version based on branch 4.2, respectively 4.2.7 -- it's been a while since v4.2.6. As usual, if there are issues not yet reported or patches to be backported, write to sr-dev mailing list. If time allows, in the same day or next days, I plan to do the last release of branch 4.1 to mark the end of official packaging in 4.1 series. At this moment, the last two stable branches are 4.3 and 4.2. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

8 years, 4 months

Presence Module

by Loïc Boissy

Hello, I use kamailio 4.3.2 with the presence module. When the contact is able to manage its presence, No problem for me. But when a contact doesn't manage it presence, i would like to know if it is possible to generate a sip message (like as a simple presence message) with kamailio? The goal of that is to prevent my watcher when a "simple" contact (without presence option) is registered and unregistered on my kamailio server. Thank you for your help. Best regards. Loïc BOISSY

8 years, 4 months

State-full failover Kamailio+RTPProxy

by Frank Costeira

Hi, Is there anyway to do state-full failover of Kamailio+RTPProxy from a primary server to a backup server? Regards, Frank

8 years, 4 months

New logo for Kamailio project

by Daniel-Constantin Mierla

Hello, based on the feedback after the proposal, there were no voices against the new logo. Therefore, as of today, Kamailio project is officially using the new logo -- see it at: - http://www.kamailio.org/w/2015/12/new-logo-for-kamailio-project/ If you display one of the old Kamailio logos, we would encourage to use the new one at your earliest convenience. That will help to propagate it faster. Posting about the change on blogs, forums or social media channels would be appreciated as well. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

8 years, 4 months

RFC: Renaming dialog_ng to ims_dialog

by Daniel-Constantin Mierla

Hello, I am under the impression that the name dialog_ng creates confusion out there and some people are using it instead of the classic dialog module. Although it was started with goals of reworking dialog module with a different concept (which was discussed mainly by some guys that afterwards changed their job to non-voip area), dialog_ng ended up to be tailored for IMS needs. Probably we should do that refactoring of the dialog module, but meanwhile dialog_ng doesn't refect that and some people are confused by the current naming of the two modules. Practically is more about convenience at this moment and if IMS developers and users think it is not going to be a big overhead for their deployments to be upgraded, I can take care to rename it. So, while general opinion matters, I think we should see first what IMS devs prefer. I am personally not affected that much, so I am fine to keep it like it is now -- in that case, proper notes should be added to documentation, stating that dialog_ng must be used only for IMS (or when the config writer knows very well what she/he is doing). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

8 years, 4 months

IMC module

by Alexandru Covalschi

Hello again First of all I wanted to ask if someone ever implemented that http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC Second question is - I don't understand the logic. In description is said: Handles Message method.It detects if the body of the message is a conference command.If so it executes it, otherwise it sends the message to all the members in the room. But why in example (well however it has broken syntax) to IMC manager are sent only messages from chat-rooms? How message from client can possibly reach imc-manager then? Also - when I send message with body "1111111" from user 1001 and imc_manager catches it - I receive 500 command error. Why? :/ All that is working on top of sipjs-demo. -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

8 years, 4 months

rtpengine and serial forking with multiple network interfaces

by Alex Balashov

Hi, It is not very clear to me how to handle serial forking scenarios with rtpengine where there are multiple RTP listeners attached to multiple network interfaces. That is to say, given: rtpengine ... -i net1/xxx.xxx.xxx.xxx -i net2/yyy.yyy.yyy.yyy 1. I attempt to reach SIP gateway 1 via net1, calling rtpengine_manage() in the process: rtpengine_manage("replace-origin replace-session-connection ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)"); 2. The call to gateway 1 fails, and I roll over to gateway 2, reached via net2, with dispatcher. I have some logic that checks how $rd has changed and sets $avp(rtp_dest) accordingly in the dispatcher failure_route[]. 3. In that failure_route[], I call: rtpengine_manage("replace-origin replace-session-connection ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)"); 4. When looking at the SDP for the INVITE on branch #2 going out to gateway 2, it does not appear to utilise media endpoint yyy.yyy.yyy.yyy, but rather continues to represent itself outwardly as xxx.xxx.xxx.xxx, even to gateway #2. I can't figure out what I'm doing wrong here, and the 'rtpengine' documentation only has one mention of serial forking, in the context of the discussion of the 'via-branch' parameter. As far as I can tell, this is not relevant to my use-case; I am not concerned about deleting the whole call from rtpengine, but rather about updating the SDP offer consonantly with serial forking to gateway #2. Any help would be deeply appreciated! Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

8 years, 4 months

Relaying failure codes back to initial server

by Alexandru Covalschi

Hello everyone! I need to relay 486/408/... other failure codes back to initial INVITE server. Here http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is recommended just to exit failure_route, but that didn't work for me. I need that to let Freeswitch know which cause has ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

8 years, 4 months

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sr-users December 2015