- sr-users - kamailio.org

Kamailio and FreeSwitch integration
by Timothy oladapo olawuyi 19 Oct '17

19 Oct '17

Dear All, I am a new entrant in Kamailio though little understanding of SIP and FreeSwitch soft switch, however, I have a project to integrate Kamailio and Freeswitch. Here is the scenario of what I want to implement; Kamailio will act as our SIP control while Freeswitch will act media server for incoming calls. Freeswitch will send all outgoing calls to Kamailio for onward transfer to our SIP provider network. No registration, presence, location Accounting, Authentication etc. are required. I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and Kamailio integration sample config available at http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still unable to figure out how to go with the configuration. I will appreciate any one who can provide guideline for achieving the above scenario. Regards Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 |Skype: daptims | <htttp://www.pethahiah.com/> htttp://www.pethahiah.com

3 4

port forwarding
by Andrzej Kaczmarczyk 19 Oct '17

19 Oct '17

Hello, This is probably trivial, but I'm quite new in kamailio and trying to handle it for my personal purposes (so - no trainings, no support etc :) ) - for few (up to 20) users and rare connections, more to learn something than to have production system. No plans to go outside my own server. I have LAN. One of machines in this LAN is server with kamailio onboard. It works, but only with local hostname (user@hostname). Now I want to go outside (with clients). As temporarty solution I have only dynamic DNS (no-ip.org) and I didn't expect it working. But tried with IP adress (user(a)WAN.IP.ADD.RESS). And... I don't know which ports to redirect on router. Now there are redirected 5060 and 10000-30000 (maybe it is not necessary but first I want it working) - taken from some on-line info. I can: - log in to all my accounts, - depends on client - call to another user and in some cases I can receive the call (so it is probably issue on my clients, because it is possible to make a call at all). But I cannot transmit any data: connections are completely silent in any case and I cannot start video - although cameras are findable. Which ports in fact are necessary? I want audio and video communication, presence info and probably nothing more. Which ports should I redirect? -- Pozdrawiam Andrzej Kaczmarczyk

2 1

[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 19 Oct '17

19 Oct '17

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2018 takes place 3-4 February 2018 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 4th of February 2018. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 30th of November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM18/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room 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. You can find the full list of dev-rooms at https://www.fosdem.org/2018/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room 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 (dev-rooms, 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 dev-room 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 dev-room administrators based on the number of received proposals. As the two previous dev-rooms 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 dev-room 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, 3rd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on the 2nd of February 2018 http://wiki.xmpp.org/web/Summit_22 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 2nd of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. 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 for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room 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 http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://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: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

1 0

Problems with AAR over Rx Interface
by Ron McLeod 19 Oct '17

19 Oct '17

When a LTE handset client attempts to REGISTER with the IMS, an AAR is sent from the P-CSCF to the PCRF to establish a dedicated bearer for SIP signaling (as well as request notifications for certain access network event). There are a couple of problems with the flow descriptions, and I was hoping that someone could confirm that these are actual issues before I start working on a fix. 1. The "in" and "out" directions seem reversed. "in" corresponds to the uplink (UE to network), and "out" corresponds to the downlink (network to UE). In the AAR sent to the PCRF, the "in" description indicates the traffic "from" the UE, rather than "to" the UE. There is a similar problem with the "out" description. A fix would involve reversing "in" and "out". 2. The IP address and port number for the UE side are correct (in my case the UE has been allocated 172.25.152.195, and the client is using port 37843), but the network side is specified as IP address 127.0.0.1 and port number is the same as the open used by the client (in my case the P-CSCF has an IP address of 172.25.152.20 and a port number of 4060). I see in the source code that the P-CSCF address is hard-coded to "127.0.0.1". A fix would involve either using the actual UE-facing IP address and port number for the P-CSCF or wildcards. Another option would be to not try and setup a dedicated bearer for SIP signaling, and just use the default bearer (there shouldn't be anything else other than signaling-related traffic on the default bearer anyway). The access network HSS would specify that the default bearer was created with the appropriate parameters (QCI=5, AM, etc.). This is the network message flow: 172.25.152.195 -> pcscf.core.ims1.test SIP 999 Request: REGISTER sip:provider1.test (fetch bindings) | pcscf.core.ims1.test -> 172.25.152.195 SIP 359 Status: 100 Trying (0 bindings) | pcscf.core.ims1.test -> pcrf.ims1.test DIAMETER 762 cmd=AARequest(265) flags=RP-- appl=3GPP Rx(16777236) h2h=28fe20b1 e2e=fe338ed9 pcrf.ims1.test -> pcscf.core.ims1.test DIAMETER 198 cmd=AAAnswer(265) flags=-P-- appl=3GPP Rx(16777236) h2h=2 e2e=13 pcscf.core.ims1.test -> 172.25.152.195 SIP 410 Status: 408 Request Timeout (0 bindings) | This is the DIAMETER AAR message: Diameter Protocol Version: 0x01 Length: 696 Flags: 0xc0, Request, Proxyable Command Code: 265 AA ApplicationId: 3GPP Rx (16777236) Hop-by-Hop Identifier: 0x28fe20b1 End-to-End Identifier: 0xfe338ed9 AVP: Session-Id(263) l=40 f=-M- val=pcscf.core.ims1.test;548405219;1 AVP: Origin-Host(264) l=28 f=-M- val=pcscf.core.ims1.test AVP: Origin-Realm(296) l=22 f=-M- val=provider1.test AVP: Auth-Application-Id(258) l=12 f=-M- val=3GPP Rx (16777236) AVP: Vendor-Specific-Application-Id(260) l=32 f=-M- AVP: Destination-Realm(283) l=22 f=-M- val=core.ims1.test AVP: Subscription-Id(443) l=52 f=-M- AVP: Media-Component-Description(517) l=304 f=VM- vnd=TGPP AVP Code: 517 Media-Component-Description AVP Flags: 0xc0 AVP Length: 304 AVP Vendor Id: 3GPP (10415) Media-Component-Description: 00000206c0000010000028af0000000100000207c00000bc... AVP: Media-Component-Number(518) l=16 f=VM- vnd=TGPP val=1 AVP: Media-Sub-Component(519) l=188 f=VM- vnd=TGPP AVP Code: 519 Media-Sub-Component AVP Flags: 0xc0 AVP Length: 188 AVP Vendor Id: 3GPP (10415) Media-Sub-Component: 000001fdc0000010000028af00000001000001fbc0000046... AVP: Flow-Number(509) l=16 f=VM- vnd=TGPP val=1 AVP: Flow-Description(507) l=70 f=VM- vnd=TGPP val=permit out ip from 172.25.152.195 37843 to 127.0.0.1 37843 AVP: Flow-Description(507) l=69 f=VM- vnd=TGPP val=permit in ip from 127.0.0.1 37843 to 172.25.152.195 37843 AVP: Flow-Usage(512) l=16 f=VM- vnd=TGPP val=AF_SIGNALLING (2) AVP: Media-Type(520) l=16 f=VM- vnd=TGPP val=CONTROL (4) AVP: Codec-Data(524) l=28 f=VM- vnd=TGPP val=downlink\noffer\n AVP: Codec-Data(524) l=27 f=VM- vnd=TGPP val=uplink\nanswer\n AVP: Flow-Status(511) l=16 f=VM- vnd=TGPP val=ENABLED (2) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=CHARGING_CORRELATION_EXCHANGE (1) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_LOSS_OF_BEARER (2) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_RECOVERY_OF_BEARER (3) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_RELEASE_OF_BEARER (4) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=INDICATION_OF_ESTABLISHMENT_OF_BEARER (now void) (5) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=IP-CAN_CHANGE (6) AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=ACCESS_NETWORK_INFO_REPORT (12) AVP: Framed-IP-Address(8) l=12 f=-M- val=172.25.152.195 AVP: Authorization-Lifetime(291) l=12 f=-M- val=7200 AVP: Auth-Grace-Period(276) l=12 f=-M- val=0 AVP: Session-Timeout(27) l=12 f=-M- val=7200

2 1

Problem removing headers from FAILURE_ROUTE
by Joel Serrano 17 Oct '17

17 Oct '17

Hi all, I have the following scenario: Client -> Kamailio (adds X-CallSource hdr) -> Media Server -> Kamailio (removes all "X-" headers) -> Carrier Carrier doesn't get any X- headers. All works. Now for example, if the media server replies with a non-OK status, like: Client -> Kamailio (adds X-CallSource hdr) <-> Media Server (480 Temporarily Unavailable) Kamailio (in failure_route, removes all "X-" headers) -> External Voicemail Service External Voicemail Service gets the X- headers. Basically, on the first example, it is a regular call, all handled from request route. On the second example, call goes to media server, media server replies with a 480, reaches kamailio's failure route, in there we run the same remove_hf_re. In the captures I can see that the X-CallSource header (added at the very beginning) is still there. I have tried different combinations in the failure route: a) Tried removing exactly the header: remove_hf("X-CallSource") b) Tried removing any X- header: remove_hf_re("X-") Those didn't work, then I tried: c) append_hf("X-CallSource: Test") remove_hf("X-CallSource") The result was now two X-CallSource headers, the add worked, the remove didn't. I thought maybe "msg_apply_changes()" could help, but it isn't valid inside failure_route. So what is the deal here? Am I doing something completely wrong or is it just that you cannot remove headers (but you can add) in failure route? According to the doc: https://www.kamailio.org/docs/modules/devel/modules/textops.html#textops.f.… -----> This function can be used from ANY_ROUTE. OS: Debian 8 Kamailio: 4.4.5 (installed from deb repo) # kamailio -v version: kamailio 4.4.5 (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 with gcc 4.9.2 Any suggestions? Thanks Joel.

2 4

Re: [SR-Users] Touching Base / Question
by Maxim Sobolev 17 Oct '17

17 Oct '17

len = snprintf(cmd->buf_t, sizeof(cmd->buf_t), "%d %lu %lu %lu %lu\n", ttl, pst[0].npkts_in, pst[1].npkts_in, pcnts.nrelayed, pcnts.ndropped) Those are: remaining session TTL in seconds, number of packets received from side A, number of packets received from side B, overall number of packets relayed in both directions (A->B, B->A) for directions, and number of packets dropped due to remote address not being known yet. There is actually a newer version of that command in the RTPproxy, which provides access to these and number of other per-session statistics collected by the analysis code, such as: rtpa_nsent rtpa_nrcvd rtpa_ndups rtpa_nlost rtpa_perrs The command to fetch those is for example: Q[v] %%CALLID%% %%TOTAG%% %%FROMTAG%% rtpa_nsent rtpa_nrcvd rtpa_ndups rtpa_nlost rtpa_perrs Full list is here: https://github.com/sippy/rtpproxy/blob/master/src/rtpp_command_query.c It would be good to get those implemented in the Kamailio as well. I would also probably hook up jitter values into that list. -Max On Mon, Oct 2, 2017 at 10:09 AM, Michael Mavroudis <michael(a)voicetel.com> wrote: > I'm sorry I should had added those details. I'm referring to: > > https://kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html > > <https://kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html> > I found a snippet at: > > http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=c > ommitdiff;h=85ee107929a80b54d47eaabef6f25d7b2f0659e7 > > There is 5 values but I'm not sure what they represent. I've also asked > Daniel so you may be hearing for him as well. > > On Sep 29, 2017 3:51 PM, "Maxim Sobolev" <sobomax(a)sippysoft.com> wrote: > >> Hi, Michael, I am not sure about October, have some other travel plans >> that are not very compatible with being in Orlando. With regards to the rtpstat, >> which counters are you using specifically? >> >> -Max >> >> On Thu, Sep 28, 2017 at 12:42 PM, Michael Mavroudis <michael(a)voicetel.com >> > wrote: >> >>> Hi, >>> >>> Hope all is well. Are you going to be in Orlando for Astricon? If not >>> I'm guessing Europe again. >>> >>> I've got a quick question for you? >>> >>> What are the contents of the psuedo var: $rtpstat >>> >>> I started calculating MOS scores and passing them to freeswitch but my >>> calculation isn't quite right. I'm thinking it's because I'm calculating >>> based on the wrong vars. Any help is greatly appreciated. >>> -- >>> Michael J. Mavroudis >>> >>> [image: VoiceTel] <http://goo.gl/yXuqFz> >>> >>> 630 Fifth Avenue >>> <https://maps.google.com/?q=630+Fifth+Avenue+New+York,+NY+10111&entry=gmail&…> >>> New York, NY 10111 >>> <https://maps.google.com/?q=630+Fifth+Avenue+New+York,+NY+10111&entry=gmail&…> >>> >>> M: (917) 426-9000 >>> W: (212) 942-7400 >>> >> >> >> >> -- >> Maksym Sobolyev >> Sippy Software, Inc. >> Internet Telephony (VoIP) Experts >> Tel (Canada): +1-778-783-0474 <%28778%29%20783-0474> >> Tel (Toll-Free): +1-855-747-7779 <%28855%29%20747-7779> >> Fax: +1-866-857-6942 <%28866%29%20857-6942> >> Web: http://www.sippysoft.com >> MSN: sales(a)sippysoft.com >> Skype: SippySoft >> > -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts Tel (Canada): +1-778-783-0474 Tel (Toll-Free): +1-855-747-7779 Fax: +1-866-857-6942 Web: http://www.sippysoft.com MSN: sales(a)sippysoft.com Skype: SippySoft

1 0

Development for Kamailio v5.1 series is frozen
by Daniel-Constantin Mierla 17 Oct '17

17 Oct '17

Hello, short note to mark the freezing of development for Kamailio v5.1 series. For few weeks, please do not push new features in the master branch. Once the branch 5.1 is created (expected to happen in 3-4 weeks from now), the master branch becomes again open for new feature. Meanwhile the focus has to be on testing current code. Work on related tools (e.g., kamctl) or documentation can still be done as well as getting the new modules in 5.1 in good shape, plus adding exports to kemi interface (which should not interfere with old code). The entire testing phase is expected to be 4 to 6 weeks, then the release of v5.1.0 -- likely by end of November should be out. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Roadmap to Kamailio v5.1
by Daniel-Constantin Mierla 16 Oct '17

16 Oct '17

Hello, being discussed at the previous IRC devel meeting before the summer, it's time to plan the preparation for next major release, respectively Kamailio 5.1 series. As an initial proposal, a timeline can be: - Mon, Oct 16, 2017 - freeze the development - in 3-4 weeks after that, create the branch 5.1 - 2-3 weeks after, do the release of v5.1.0 That means more or less by end of November we will have first version of 5.1 series. There are still 4 weeks of development, hopefully enough to get ongoing work finished. I know from github tracker that there are people working on redis support for usrloc. I will focus on exporting more functions to kemi, right now should be around 60%, by freeze should be more than 80% (likely that means all of what most of people use these days, because there are some modules that I guess are not much used). Maybe one or two modules will be pushed as well from my side. I think we should also do an IRC devel meeting to try to sync better on this. Maybe by mid of next week, I will send another email shortly. Should anyone have more details to add or adjust the proposal, do not hesitate to reply to mailing list! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 2

Having kamailio bind to a VRF device
by George Diamantopoulos 15 Oct '17

15 Oct '17

Hello, I have a use case where I need to have kamailio bind to a VRF device. The configuration in question is similar to the example below, where eth1 is a slave to the VRF-lite device: +----------+ +-------------------+ | eth0 | | vrf-green | | 1.1.1.1 | | 127.0.0.1 | +----------+ +-------------------+ | +----------+ | eth1 | | 2.2.2.2 | +----------+ Both the main routing table and "vrf-green" routing table have a default route. What I need to be able to do is have kamailio bind to both interfaces: listen=eth0:5060 listen=vrf-green:5060 And additionally be able to use force_send_socket to select an interface, for example: force_send_socket(udp:2.2.2.2:5060); However, I can't get this to work. The above configuration fails because there is no listen directive for 2.2.2.2. Also, kamailio doesn't process packets received on the VRF with the above listen directives, it behaves as if it doesn't listen on 2.2.2.2 indeed. In addition using either of the below: listen=udp:2.2.2.2:5060 or listen=eth1:5060 fails with an error upon starting kamailio. According to the kernel documentation: Applications that are to work within a VRF need to bind their socket to the VRF device: setsockopt(sd, SOL_SOCKET, SO_BINDTODEVICE, dev, strlen(dev)+1); or to specify the output device using cmsg and IP_PKTINFO. The question is, is VRF useable with kamailio right now? Or is development needed? Thanks! BR, George

5 15

rtcp-mux policy for RTPengine module
by Yuriy Gorlichenko 14 Oct '17

14 Oct '17

Hi all! Some time ago Chromium browser sets rtcpMuxPolicy: required by default (soon on Chrome 58) It means that webRTC based clients not accepts a=rtcp:31757 And uses for RTP and RTCP multiplexing at one port Main trouble that i found: calls between original SIP client and webRTC client (SIP client is initiator of call) When sip client sends invite it has a=rtcp:33445 Means it wants 2 different prots for RTCP and RTP As expected for this case webRTC client says 488 Not accessible here instead of 200 resonse I suppose rtpengine module should hept to handle it but i can not find any key how to do it I added form rtpengine_manage() rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux" But not removes a=rtcp and ice cadidate prepeared for it. Suppose removing a=rtcp:12345 will gives just an issue for RTP session. Does rtpengine module have some keys for resole this issue?

2 3

It does not log any route. Clean installation
by Bruno Henrique Gomes Gouvêa 14 Oct '17

14 Oct '17

Hello, I am doing some tests with version 4.2.3 (installed of RPM centos 7 packages) in a clean installation, I have a kamailio running in production already configured and working, but I would like to start a totally clean configuration to learn more about kamailio. I am using the configuration file that comes in the default installation (kamailio-basic.cfg) just by adding a log on the main route. I kept the log in messages file and increased the verbosity as below, and when i send any package (INVITE or REGISTER) I assume that the log or xlog lines should be displayed in file, but no. I see the packets arriving via tcpdump, and netstat shows that port 5060 is being listened by kamailio, but I have nothing else besides the initialization information (and the debug of MYSQL querys when activated unrelated to INVITE or REGISTRY packets). Am I doing something wrong? *Configs **Log :* ### LOG Levels: 3 = DBG, 2 = INFO, 1 = NOTICE, 0 = WARN, -1 = ERR #! ifdef WITH_DEBUG debug = 4 log_stderror = no #! else debug = 4 log_stderror = no #! endif memdbg = 5 memlog = 5 log_facility = LOG_LOCAL0 *Main route log test:* request_route { log ("Log main request TEST"); xlog ("Log main request TEST 2"); # not displayed route (REQINIT); */var/log/messages:* Oct 5 23:39:12 pabxmodelo kamailio: Stopping kamailio: [ OK ] Oct 5 23:39:12 pabxmodelo systemd: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Oct 5 23:39:13 pabxmodelo kamailio: INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected) Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30050]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Oct 5 23:39:13 pabxmodelo kamailio: Starting kamailio: Oct 5 23:39:13 pabxmodelo /usr/sbin/kamailio[30070]: INFO: ctl [io_listener.c:225]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Oct 5 23:39:13 pabxmodelo systemd: Started SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Oct 5 23:39:13 pabxmodelo kamailio: [ OK ] -------------------------------------------- Bruno H. G. Gouvêa ---

2 4

Re: [SR-Users] DIAMETER over SCTP for Rx and Cx Interfaces
by Carsten Bock 13 Oct '17

13 Oct '17

Hi, I am working on an update, to support SCTP for the CDP module. At the moment, I have an incomplete prototype, which binds to the Kamailio sockets (so any protocol Kamailio supports) to send/receive Diameter messages. I plan to finish it after my holiday, so I might have a full working prototype for TADsummit in Lisbon mid November. Thanks, Carsten Am 13.10.2017 14:00 schrieb "Ron McLeod" <ron.kamailio(a)mcleodnet.com>: I would like to connect Kamailio to a PCRF (Rx interface) and HSS (Cx interface) using DIAMETER over SCTP. Is this currently supported? If it is, are there any notes on how to configure this? If not, and there are plans to develop this, I would like to get involved in testing. Thanks. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 1

Handle SDP ICE candidates
by Thiago Guimarães 13 Oct '17

13 Oct '17

Hello, I would like to know if there is any specific configuration in kamailio.conf to enable/disable handle the array of ICE candidates. In my tests, using PJSIP for the development of my User Agent, I notice that when ICE is enabled, several calls made do not reach the other User Agent and in this case they receive a 408 error. I am suspicious that Kamailio is getting lost, when the SDP arrives with the array of ICE candidates. The initial goal of using ICE is to promote peer-to-peer communication when both are under the same router, because the application is always with STUN enabled. Best regards Thiago Barcelos Guimarães thiago.barcelos(a)gmail.com

3 4

DIAMETER over SCTP for Rx and Cx Interfaces
by Ron McLeod 13 Oct '17

13 Oct '17

I would like to connect Kamailio to a PCRF (Rx interface) and HSS (Cx interface) using DIAMETER over SCTP. Is this currently supported? If it is, are there any notes on how to configure this? If not, and there are plans to develop this, I would like to get involved in testing. Thanks.

1 0

custom kamailio-x.deb package
by Abdul Basit 13 Oct '17

13 Oct '17

Hi All, I am trying to compile custom .deb package from kamailio source that will install dependencies like mysql-server, lua, rtpproxy, etc. and will install few custom scripts along with additional db tables from tables-create.sql files. I am able to create kamailio_x-amd64.deb package that I installed on fresh machine dpkg -i kamailio_x-amd64.deb This package installed kamailio config files and modules in appropriate locations but it didn't created default, init.d and kamailio.service files. Nor it asked for installing the dependencies which I mentioned in control file. OS: Debian GNU/Linux 8.9 (jessie) Kamailio: 5.0.3 through git clone I am creating .deb package for the very 1st time. Can anyone point me to the right direction? -- regards, abdul basit

3 5

How to troubleshoot "ERROR: error while initializing modules"
by ron.kamailio＠mcleodnet.com 13 Oct '17

13 Oct '17

I have built kamailio-5.0.3 from source with no issues. I have used the P-CSCF configuration examples from ./misc/examples/pcscf and the database definitions from ./utils/kamctl/mysql. When I check the configuration file, it is reported to be "ok": # /usr/local/sbin/kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(10835) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(10835) INFO: <core> [core/sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* config file ok, exiting... But when I attempt to run the application, it fails with an error message "ERROR: error while initializing modules": # /usr/local/sbin/kamailio -D -d loading modules under config path: /usr/local/lib64/kamailio/modules/ Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* WARNING: no fork mode ERROR: error while initializing modules How can I troubleshoot this to determine what the actual error is? I have tried increasing the debug level, but I don't see any additional information. Thanks.

2 1

Re: [SR-Users] How to troubleshoot "ERROR: error while initializing modules" - RESOLVED
by ron.kamailio＠mcleodnet.com 12 Oct '17

12 Oct '17

More information was available in /var/log/messages: Oct 12 15:23:25 p-cscf kamailio: INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 Oct 12 15:23:25 p-cscf kamailio: ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Oct 12 15:23:25 p-cscf kamailio: ERROR: ctl [ctl.c:290]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Oct 12 15:23:25 p-cscf kamailio: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) I resolved the problem by providing the proper path for these two configuration items: modparam("ctl", "binrpc", "unix:/var/run/kamailio_pcscf/kamailio_ctl") modparam("dispatcher", "list_file", "/etc/kamailio_pcscf/dispatcher.list") -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of ron.kamailio(a)mcleodnet.com Sent: Thursday, October 12, 2017 1:36 PM To: 'Kamailio (SER) - Users Mailing List' Subject: [SR-Users] How to troubleshoot "ERROR: error while initializing modules" I have built kamailio-5.0.3 from source with no issues. I have used the P-CSCF configuration examples from ./misc/examples/pcscf and the database definitions from ./utils/kamctl/mysql. When I check the configuration file, it is reported to be "ok": # /usr/local/sbin/kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(10835) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(10835) INFO: <core> [core/sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* config file ok, exiting... But when I attempt to run the application, it fails with an error message "ERROR: error while initializing modules": # /usr/local/sbin/kamailio -D -d loading modules under config path: /usr/local/lib64/kamailio/modules/ Listening on udp: pcscf.core.ims1.test [172.25.152.20]:4060 Aliases: *: pcscf.core.ims1.test:* WARNING: no fork mode ERROR: error while initializing modules How can I troubleshoot this to determine what the actual error is? I have tried increasing the debug level, but I don't see any additional information. Thanks. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

presentce handly_subscribe() protocol/port mismatch to WS endpoint
by Yuriy Gorlichenko 12 Oct '17

12 Oct '17

Hi. I have presence server and it works fine for UDP/TCP/TLS endpoints. For now i have new one type of endpoints that runs via WebSockets It sends SUBSCRIBE request to the and then after handle_subscribe() NOTIFY not comes to the subscriber because of [core/forward.c:231]: get_send_socket2(): protocol/port mismatch I already had some issues regarding this for ACK for example but i resolved it cimply doing $ru = $ru+";transport=wss" but NOTIFY sending is internal process and can't be controlled by config file. So i can not change $ru for NOTIFY directly. Any ideas how to fix this?

2 18

Inserting local path into shared USRLOC table.
by Marrold 12 Oct '17

12 Oct '17

Hi All, I am wondering if the following is possible. For simplicity imagine two asterisk boxes and two kamailio proxies. Kamailio on the edge, asterisk in the 'core' Kamailio <=> Asterisk Asterisk <=> Kamailio A user may register to either kamailio proxy, and a call may originate from either Asterisk box where it will be sent to its *local *Kamailio proxy. >From there I'd like Kamailio to forward the request to the AoR, following the same path the registration took if it registered to the other proxy, to avoid NAT issues. I think this requires a shared USRLOC table, and also for the proxy that handled the registration to insert itself in the path of the USRLOC table. The registrar module has the 'path_check_local' parameter which sounds relevant, ignoring the path if the first hop is local. I've included add_path_received() in my REGISTRAR routing block, however I see 'Path: [not set]' in the USRLOC table. Is it possible to force the insertion of the local proxy into the path field? One additional question, in the case of NAT keepalives, is it possible to only send these from the proxy the received the register request? Thanks in advance, Matthew

4 4

kamailio QOS questions
by Lv, Victor (NSB - CN/Qingdao) 12 Oct '17

12 Oct '17

Hi All, Currently, we would like to have a try on the Kamailio QOS module toward PCRF Rx interface, it's looks a little complex to configure. Is there anyone had already successfully configured the Rx interface? If yes, could you share your configuration with me? Thanks in advance. Victor

3 3

Forcing a TCP Connection Closed?
by Mark Boyce 11 Oct '17

11 Oct '17

Hi all Just working on some connections security filters on a Kamailio install. The security goes something like this; In REQINT … if source_ip is not in customers IP white-list then just exit This works fine for UDP where packets are just ignored if they don’t come from a trusted IP. However on TCP this leads to the connection staying open until it either times out or the source disconnects. Which feels untidy. Is there a way to say close the TCP connection from within the config script? Thanks Mark -- Mark Boyce

3 5

Route Header - Comma separated issue
by Soltanici Ilie 11 Oct '17

11 Oct '17

Hi all, Why Kamailio are not removing full Route header in case if he is receiving comma separated Route? For example: Receiving such Message: SUBSCRIBE sip:101@192.168.1.1 SIP/2.0 Route: <sip:192.168.1.1;r2=on;lr=on;ftag=fb0076047b3da3e4;nat=yes>, <sip:172.16.30.10;sbc=topoh-NFlx.VXxPVXv8rfM83X1KnhF83LINVLO8VcJ6VL79zkEYH5U83hdKVhyKFi-5zX-5njs67kyYSllNx**> Event: dialog User-Agent: Cisco/SPA504G-7.6.2 Content-Length: 0 after applying loose_route function, kamailio is forwarding this message: SUBSCRIBE sip:101@192.168.1.1 SIP/2.0 Route: Event: dialog User-Agent: Cisco/SPA504G-7.6.2 Content-Length: 0 Other devices are sending 2 Route Headers in separate rows - and everything is working properly. Is this a bug or it should be like this per RFC? Thank You. using Kamailio 5.0.3.

1 0

[4.2.8] SUBSCRIBE message was not handled
by Jack Wang 11 Oct '17

11 Oct '17

Hello everyone, According to the routing flow set in kamailio.cfg # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); After I traced the flow it seems that SUBSCRIBE message failed on t_check_trans() and stopped there. I add some logs to keep tracing this function and found that: int t_check_trans(struct sip_msg* msg) { struct cell* t; int branch; int ret; /* already processing a T */ if(is_route_type(FAILURE_ROUTE) || is_route_type(BRANCH_ROUTE) || is_route_type(BRANCH_FAILURE_ROUTE) || is_route_type(TM_ONREPLY_ROUTE)) { return 1; } if (msg->first_line.type==SIP_REPLY) { branch = 0; ret = (t_check_msg( msg , &branch)==1) ? 1 : -1; tm_ctx_set_branch_index(branch); return ret; } else if (msg->REQ_METHOD==METHOD_CANCEL) { return w_t_lookup_cancel(msg, 0, 0); } else { switch(t_check_msg(msg, 0)){ case -2: /* possible e2e ack */ return 1; case 1: /* found */ t=get_t(); if (msg->REQ_METHOD==METHOD_ACK){ /* ack to neg. reply or ack to local trans. => process it and end the script */ /* FIXME: there's no way to distinguish here between acks to local trans. and neg. acks */ if (unlikely(has_tran_tmcbs(t, TMCB_ACK_NEG_IN))) run_trans_callbacks(TMCB_ACK_NEG_IN, t, msg, 0, msg->REQ_METHOD); t_release_transaction(t); } else { /* is a retransmission */ if (unlikely(has_tran_tmcbs(t, TMCB_REQ_RETR_IN))) run_trans_callbacks(TMCB_REQ_RETR_IN, t, msg, 0, msg->REQ_METHOD); t_retransmit_reply(t); } /* no need for UNREF(t); set_t(0) - the end-of-script t_unref callback will take care of them */ return 0; /* exit from the script */ <---------------------------- THE POINT !! } /* not found or error */ } return -1; } If the line "return 0; /* exit from the script */" was changed to "return 1; /* exit from the script */" , it works ---- means that the configuration script can keep being proceeded now. Any suggestions?

2 4

200 OK - need to include Expires
by Subba Krishna MG 11 Oct '17

11 Oct '17

Hello, This might be very simple question for you folks, I am using Kamailio as SIP - Presence server, and For every user Kamailio is sending 200 OK, but the EXPIRES IE is missing. Can u help me include this IE or any other IE for 200 OK message? How to modify the configuration files to get this done? Thanks a lot, MGSK

2 4

Path module adds either received or outbound
by Rick Barenthin 11 Oct '17

11 Oct '17

Hey, Maybe someone from the list can share some information about the path module. Why can the path module only add either the received parameter to path or the flow token with the ob parameter, but not both. Does the RFC mandates it? I didn't find anything like that and was wondering because the source calculates max length of the parameters with following: >> /* maximum possible length of suffix */ >> suffix_len = strlen(";lr;received=sip::12345%3Btransport%3Dsctp;ob;>\r\n") + IP_ADDR_MAX_STR_SIZE + 2 + (add_params ? add_params->len : 0) + 1; This looks like it can be possible to have both. Best regards Rick

2 1

how to use SIP header Route
by anil kumar k 11 Oct '17

11 Oct '17

HI, How to use SIP header Route in [SIPOUT] route instead of uri -- Regards, K.Anil Kumar

2 1

Route Not accessable
by anil kumar k 11 Oct '17

11 Oct '17

Dear Kamailio Uers, IN my [SIPOUT] by default not getting the Route from header . Any bode can help on this -- Regards, K.Anil Kumar

2 1

tsilo ts_store() causes segfault if method is not INVITE
by Anthony Alba 10 Oct '17

10 Oct '17

I was modifying the routing blocks of the tsilo push notifications recipe from Federico Cabiddu and could cause a segfault in kamailio 5.0.3 if due to misconfiguration an ACK reached ts_store(). ## modify https://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu… ## various route blocks will lead to here ## including ACK and INVITE... route[LOCALRELAY) { ## ooops: did not check method is INVITE t_relay(); ts_store(); $sht(vtp=>stored::$rU) = 1; xlog("L_INFO", "stored transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); ## saw an ACK and caused segfault } If properly checked for INVITE: if (is_method("INVITE")) { /* do tsilo stuff */ } it works. I'm wondering whether ts_store() should/could be more defensive and show an error like "..not the right context for ts_store().." . Thanks for you thoughts. Anthony

2 2

kamailio with NAT IP and DNS loadbalance
by mlwu＠south-star.com.cn 10 Oct '17

10 Oct '17

Hi, Here are same problems in my configuration of kamailio server. I have two kamailio server (both have a NAT IP and a lan IP), and I use DNS to keep loadbalance.When sending message to another account , if NAT IP which is returned by DNS, is not the same as receiver account, kamailio server sends 404 not found error. I really have no idea how to set the configuration when using DNS loadbalance to return two NAT IP. Thank you , I'm looking forward from you! Sincerely yours Ada WU mlwu(a)south-star.com.cn

1 1

kamailio with NAT IP and DNS loadbalance
by 吴妙灵 10 Oct '17

10 Oct '17

Hi, Here are same problems in my configuration of kamailio server. I have two kamailio server (both have a NAT IP and a lan IP), and I use DNS to keep loadbalance.When sending message to another account , if NAT IP which is returned by DNS, is not the same as receiver account, kamailio server sends 404 not found error. I really have no idea how to set the configuration when using DNS loadbalance to return two NAT IP. Thank you , I'm looking forward from you! Sincerely yours Ada WU

1 0

INFO: qq.com domain was blocked due to very high volume of spam
by Daniel-Constantin Mierla 09 Oct '17

09 Oct '17

Hello, writing to inform everyone that currently the qq.com domain is blacklisted by the kamailio.org email server due to high volume of spam sent to many admin addresses and mailing lists. Spamassassin had very hard time digesting and filtering, as content kept changing and addresses were random. I didn't get the chance yet to check if we have subscribers with this domain, so, as a fast lane, if you use it and don't have the possibility to quickly switch to another email server domain, write me to try to whitelist the address. Hopefully, if anyone is affected, there is not much inconvenience, but it was the only quick solution that could be applied for the moment. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Set music on hold
by Fedor 08 Oct '17

08 Oct '17

i got in my kamailio.cfg: loadmodule "mohqueue.so" # --- music on hold moh ---- modparam ("mohqueue", "db_url", "mysql://kamailio:PASSWORDHERE@localhost /kamailio") modparam ("mohqueue", "mohdir", "/etc/kamailio/moh") modparam ("mohqueue", "moh_maxcalls", 3000) databases for mohqueue exist fed@vps1:/etc/kamailio# ll /etc/kamailio/moh/ total 185424 drwxr-xr-x 2 kamailio kamailio 4096 Oct 8 09:32 ./ drwxr-xr-x 3 root root 4096 Oct 8 10:04 ../ -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.0 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.1 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.10 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.11 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.12 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.13 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.14 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.15 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.16 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.17 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.18 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.19 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.2 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.20 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.3 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.4 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.5 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.6 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.7 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.8 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.9 -rw-r--r-- 1 kamailio kamailio 8627660 Jan 2 2017 hold.wav rtpproxy configured, but no music on hold(srtp works fine) Can anybody help me, please?

2 1

Re: [SR-Users] How to do replace_body_all with Kemi python version kamailio 5.0.x
by Muhammad Zaka 06 Oct '17

06 Oct '17

Hi Daniel, Unfortunately, it didn't help as the "self" pointer is null in the following code at python_msgobj.c:175 if (self->msg == NULL) { PyErr_SetString(PyExc_RuntimeError, "self->msg is NULL"); Py_INCREF(Py_None); return Py_None; } Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f688ae37009 in msg_call_function (self=0x0, args=0x7f68786b7aa0) at python_msgobj.c:175 175 python_msgobj.c: No such file or directory. (gdb) bt #0 0x00007f688ae37009 in msg_call_function (self=0x0, args=0x7f68786b7aa0) at python_msgobj.c:175 #1 0x00007f688a9f0e6c in PyEval_EvalFrameEx () from /usr/lib/x86_64-linux-gnu/libpython2.7.so.1.0 #2 0x00007f688aa64190 in PyEval_EvalCodeEx () from /usr/lib/x86_64-linux-gnu/libpython2.7.so.1.0 #3 0x00007f688a9b332c in ?? () from /usr/lib/x86_64-linux-gnu/libpython2.7.so.1.0 #4 0x00007f688a9bbbe3 in PyObject_Call () from /usr/lib/x86_64-linux-gnu/libpython2.7.so.1.0 #5 0x00007f688a9fe0e5 in ?? () from /usr/lib/x86_64-linux-gnu/libpython2.7.so.1.0 #6 0x00007f688a9bbbe3 in PyObject_Call () from /usr/lib/x86_64-linux-gnu/libpython2.7.so.1.0 #7 0x00007f688aa636e7 in PyEval_CallObjectWithKeywords () from /usr/lib/x86_64-linux-gnu/libpython2.7.so.1.0 #8 0x00007f688ae32eac in apy_exec (_msg=0x7f688f3d15c8, fname=0x7f688ae37dc0 "ksr_request_route", fparam=0x0, emode=1) at python_exec.c:143 #9 0x00007f688ae136f6 in sr_kemi_config_engine_python (msg=0x7f688f3d15c8, rtype=1, rname=0x0, rparam=0x0) at apy_kemi.c:53 #10 0x0000000000579b6c in receive_msg ( buf=0xa395c0 <buf> "INVITE sip:+441992878768@185.28.212.141:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 185.28.212.125:5060;branch=z9hG4bK-15930-1-0\r\nFrom: anonymous <sip:anonymous@185.28.212.141>;tag=15930Tag001\r\nTo: <sip:+441992878"..., len=858, rcv_info=0x7fff0791c6e0) at core/receive.c:260 #11 0x00000000004a86f9 in udp_rcv_loop () at core/udp_server.c:515 #12 0x0000000000422d08 in main_loop () at main.c:1623 #13 0x000000000042a408 in main (argc=13, argv=0x7fff0791cb88) at main.c:2643 Do we have any timelines when Kamailio 5.1 will be available? Thanks Regards Muhammad Zaka From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 27 September 2017 10:06 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Muhammad Zaka <muhammad.zaka(a)cloudcall.com<mailto:muhammad.zaka@cloudcall.com>> Subject: Re: [SR-Users] How to do replace_body_all with Kemi python version kamailio 5.0.x Hello, the version 5.0 doesn't have too many exports to kemi. In master branch there is already subst_body(): - https://github.com/kamailio/kamailio/blob/master/src/modules/textops/textop… I will add more as I get time before 5.1 release. Anyhow, even in 5.0 is a generic kemi function that allows executing any kamailio.cfg function, respectively: KSR.x.modf(func, param1, ...) In your case, try: KSR.x.modf("replace_body_all", "a=sendonly", "a=sendrecv") You should be careful with KSR.x.modf(...), because is some cases it may result in pkg leak due to cfg param fixup. So, do some tests and be sure that pkg memory stats are not indicating a leak. Cheers, Daniel On 26.09.17 10:48, Muhammad Zaka wrote: Hi We are experiencing issue with python script KSR. textops.replace_body_all using branch 5.0.2: - Example Code: KSR.textops.replace_body_all("a=sendonly", "a=sendrecv") It looks like that the module textops and textopsx are not available. Using $rb or $msg(body) pseudo variables isn't working as they are readonly. Is there any other way to replace the body? Many Thanks Regards Muhammad Zaka _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - www.asipto.com<http://www.asipto.com> Kamailio World Conference - www.kamailioworld.com<http://www.kamailioworld.com>

1 0

[SOLVED] Re: rtpengine_manage in tsilo INVRESUME sends delete instead of offer
by Anthony Alba 06 Oct '17

06 Oct '17

Thank you - that explains it! On Fri, Oct 6, 2017 at 11:31 PM, Federico Cabiddu <federico.cabiddu(a)gmail.com> wrote: > Hi, > http://www.kamailio.org/docs/modules/devel/modules/rtpengine.html#rtpengine…. > If you are calling it after resuming a transaction you are basically in a > FAILURE_ROUTE, so it will call the "delete" command. You have to explicitly > call rtpengine_offer for this to work. > > Cheers, > > Federico > > On Fri, Oct 6, 2017 at 5:23 PM, Anthony Alba <ascanio.alba7(a)gmail.com> > wrote: >> >> Hi, >> >> I am using tsilo and push notification recipe from >> >> https://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu… >> >> On resume of a suspended transaction rtpengine_manage() is sending >> delete instead of offer. >> >> In the routing block INVRESUME after t_continue(): >> >> >> route[INVRESUME] { >> lookup("location"); >> rtpengine_manage("DTLS=off ICE=remove"); >> t_relay(); >> ts_store(); >> $sht(vtp=>stored::$rU) = 1; >> } >> >> I expect rtpengine_manage() to send offer since this is a Request+SDP; >> instead it sends delete to rtpengine with garbage Call-ID. If I >> replace rtpengine_manage() with rtpengine_offer() then it works. >> >> rtpengine (the actual media proxy) sees >> >> Received command 'delete' from 192.168.1.755:49851 >> Call-ID to delete not found >> >> which is totally bogus of course. >> >> rtpengine_offer() works fine but wondering why rtpengine_manage() goes >> berserk here. >> >> Anthony >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >

1 0

rtpengine_manage in tsilo INVRESUME sends delete instead of offer
by Anthony Alba 06 Oct '17

06 Oct '17

Hi, I am using tsilo and push notification recipe from https://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu… On resume of a suspended transaction rtpengine_manage() is sending delete instead of offer. In the routing block INVRESUME after t_continue(): route[INVRESUME] { lookup("location"); rtpengine_manage("DTLS=off ICE=remove"); t_relay(); ts_store(); $sht(vtp=>stored::$rU) = 1; } I expect rtpengine_manage() to send offer since this is a Request+SDP; instead it sends delete to rtpengine with garbage Call-ID. If I replace rtpengine_manage() with rtpengine_offer() then it works. rtpengine (the actual media proxy) sees Received command 'delete' from 192.168.1.755:49851 Call-ID to delete not found which is totally bogus of course. rtpengine_offer() works fine but wondering why rtpengine_manage() goes berserk here. Anthony

2 1

rtpengine - optional srtp
by Daniel-Constantin Mierla 06 Oct '17

06 Oct '17

Hello, wondering if anyone has a quick answer to spare some time searching the web or source code -- is there a way to offer optional SRTP (just SDES is also fine) in SDP with RTPEngine? The use case is when the target device is not yet known to support (or not) SRTP. There are couple of ways in specs or practices (e.g., RFC5939/6871, or advertising crypto attributes on RTP/AVP stream, or offering two streams one RTP and one SRTP), I am looking to find if it is possible to do it with RTPEngine and which of the options are supported/what are the parameters for doing it... Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 3

SIP -
by Subba Krishna MG 06 Oct '17

06 Oct '17

Hello, How do i include Expires header for 200 OK message? One of the manual says this.. how do i modify my existing config file to include this? I am using : /usr/local/etc/kamailio/kamailio.cfg for configuration. Rgds, MGSK 1.4.9. expires_offset (int) The value that should be subtracted from the expires value when sending a 200OK for a publish. It is used for forcing the client cu send an update before the old publish expires. *Default value is "0".* *Example 1-9. Set expires_offset parameter* ... modparam("presence", "expires_offset", 10)

3 5

Re: [SR-Users] Subscribe Reinvites totag
by Soltanici Ilie 06 Oct '17

06 Oct '17

I found that the problem is because of this header:

1 1

A "success_route" as counterpart to "failure_route" available?
by Kristijan Vrban 05 Oct '17

05 Oct '17

Hello, i need a "success_route" which does the same as "failure_route", but only for positiv 2xx reply. Is something like this available? Background: i forward REGISTER to external registrar server, and only what to save the location on positiv reply to REGISTER. Without doing the authentication myself. I tried to save the location in onreply_route, but then the contact of the external registrar server is saved. So i need a status check in the REQUEST_ROUTE. Best Kristijan

3 6

call disconnects after 32 seconds
by Adesh Pandey 05 Oct '17

05 Oct '17

Hi, I have set up the Kamailio as load balancer using dispatcher module but all the calls get disconnected after 30 seconds (call is forwarded to Asterisk Box and answered as well). I have used example configuration from the below URL: http://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#idp51014772 Please help me to understand what is wrong and what shall I do it to fix. Thank You!! Adesh P. -- Adesh Pandey Sr. Software Developer [image: phone] +91 92129 92129 Ext: 56 [image: moible] +91 8527384897 [image: Profile Pic] [image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter] <https://twitter.com/MyOperator>[image: Twitter] <https://www.youtube.com/user/MyOperatorCo>[image: Linkedin] <https://www.linkedin.com/company/voicetree-technologies>[image: Google+] <https://plus.google.com/+MyoperatorCo/posts> [image: Logo] <http://myoperator.co/>

3 4

Parse Error (event_route)
by James DePhillip II 04 Oct '17

04 Oct '17

I am trying to setup the internal XCAP service and kamailio will not start. The messages log states "kamailio: CRITICAL: <core> [cfg.y:3372]: yyerror_at(): parse error in config file //etc/kamailio//kamailio.cfg, line 1009, column 1-11: syntax error" This is the line "event_route[xhttp:request] {" from the config file. I am not sure why this is happening any ideas? Thanks,Jim

1 0

issue when trying to start kamailio using db_postgres and dialplan modules
by David Escartín 04 Oct '17

04 Oct '17

hello all we are having an issue when trying to start kamailio using these modules and a kamailio version higher than 4.2 we install libpq5 and libpq-dev and we always have this error 0(10800) ERROR: <core> [core/sr_module.c:582]: load_module(): could not open module </usr/local/kamailio-5.0/lib64/kamailio/modules/db_postgres.so>: /usr/local/kamailio-5.0/lib64/kamailio/modules/db_postgres.so: undefined symbol: PQconnectdbParams do you know which lib or packet would we miss? we are trying to install it on centos6, ubuntu 10 or centos7 best regards david

2 3

Re: [SR-Users] Manipulate From URI username
by Gertjan Wolzak 04 Oct '17

04 Oct '17

Thank you guys. I am using the s.substring solution, as the uac_replace is done in another route part. Rgds, Gertjan

1 0

Manipulate From URI username
by Gertjan Wolzak 04 Oct '17

04 Oct '17

Hello Kamailions, Im trying to manipulate the username from the from uri, but not having any luck, I must be doing something wrong. I receive a from uri username that is prefixed with 888, now I am trying to remove the triple 8. I have configured the following: line 1090 if($fU=~"^888"){ 1091 strip(3); 1092 } 1093 route(GETGROUP); But I do not get any result, the debug shows the following: 29(9610) exec: *** cfgtrace:request_route=[MOMANX] c=[/usr/local/etc/kamailio/kamailio.cfg] l=1093 a=16 n=if 29(9610) exec: *** cfgtrace:request_route=[MOMANX] c=[/usr/local/etc/kamailio/kamailio.cfg] l=1091 a=45 n=strip 29(9610) exec: *** cfgtrace:request_route=[MOMANX] c=[/usr/local/etc/kamailio/kamailio.cfg] l=1093 a=5 n=route Does anyone see what I am doing wrong? Help would be appreciated. Rgds, Gertjan Wolzak

4 3

"python_exec" not work
by ‪mahsa _a_‬ ‪ 03 Oct '17

03 Oct '17

Iwant to work with "app_python" module,i set “script_name”,“mode_init_function”, “child_init_function” and “python_exec”in kamailio.cfg ,but I have a problem in “python_exec”section. This section not work in correctly, please guide me.thanks

1 0

Debian Stretch
by Jurijs Ivolga 03 Oct '17

03 Oct '17

Hi Guys, I recently tried to use Debian Stretch in production and it didn't went well. On load(500k-700k SIP messages per day) I get a problem that at some point there was a pike of CLOSE_WAIT connections(up to 2k of CLOSE_WAIT connections) and no new connections was possible, I tried to figure out where problem is, but I didn't found anything in system or Kamailio logs, Kamailio just stopped to receive traffic via TCP, but UDP continued to work. Test environment where was no load I didn't faced such issue. Just curious is there anybody who are using Debian Stretch in production without issues? Jurijs

5 7

event_route[usrloc:contact-expired] not called after contact expires
by Yuriy Gorlichenko 02 Oct '17

02 Oct '17

Hi I added this eventroute for manipulate temprorary entires in my subscriber table but it is not called in my cfg. May be i didn't setted up in scf but didnf found any special params for this. I use kamailio 5.0.3 version: kamailio 5.0.3 (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 09:39:55 Sep 8 2017 with gcc 4.9.2 My registrar and usrloc modules setted up as usual modparam("registrar", "method_filtering", 1) modparam("registrar", "max_expires", 3600) modparam("registrar", "gruu_enabled", 0) modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) thx for advice

2 2

sigcomp usage with kamailio
by Ram 02 Oct '17

02 Oct '17

Hi, I need to enable sigcomp with kamailio. Can any one guide me how proceed with that. And its ok to integrate any third party tools like libsigcomp for the same. Regards Raman

2 1

Kamailio htable mod-init route: Main process exited before writing to pipe
by George Diamantopoulos 02 Oct '17

02 Oct '17

Hello all, I've been trying to utilise event_route[htable:mod-init] in order to load information from the database into an htable on startup. However, Kamailio exits with an error for no obvious (to me) reason. Here's the log output: DEBUG: htable [htable.c:226]: child_init(): rank is (-127) DEBUG: htable [htable.c:259]: child_init(): executing event_route[htable:mod-init] (2) DEBUG: <core> [core/parser/msg_parser.c:600]: parse_msg(): SIP Request: DEBUG: <core> [core/parser/msg_parser.c:602]: parse_msg(): method: <OPTIONS> DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): uri: < sip:you@kamailio.org> DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): version: <SIP/2.0> DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=2 DEBUG: <core> [core/parser/msg_parser.c:491]: parse_headers(): Via found, flags=2 DEBUG: <core> [core/parser/msg_parser.c:493]: parse_headers(): this is the first via ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe And here's the routing block in kamailio.cfg. What I need to do is get the username for all non-expired contacts in the "location" table, perform a query on another database for each one of them and store the result in an htable: event_route[htable:mod-init] { sql_pvquery("ck", "select username from location where now() <= expires", "$avp(registered_ng)"); $var(i) = 0; while ( $(avp(registered_ng)[$var(i)]) != $null ) { sql_pvquery("ca", "select property from tablename where username = $(avp(registered_ng)[$var(i)])", "$var(ng_property)"); $sht(isnguser=>$var(ng_property)) = 1; $var(i) = $var(i) + 1; pv_unset("$var(ng_property)"); } pv_unset("$var(i)"); } Any hints as to what I might be doing wrong? Thanks! Best regards, George

2 5

Regarding ASYNC module in kamailio - handling SIP replies
by Rajagopal Maddi 02 Oct '17

02 Oct '17

Hi, We are checking the kamailio source code for the ASYNC module. The system is working for REQUEST_ROUTE, but not for REPLY_ROUTE. We added the ANY_ROUTE in the cmd_export_t, but still the async_route is not able to suspend for SIP Reply (For example, 401 - Unauthorized). Can we get some help on how to perform this. Some hints on how to perform would really help our development. static cmd_export_t cmds[]={ {"async_route", (cmd_function)w_async_route, 2, fixup_async_route, 0, REQUEST_ROUTE|FAILURE_ROUTE|ANY_ROUTE}, {"async_sleep", (cmd_function)w_async_sleep, 1, fixup_async_sleep, 0, REQUEST_ROUTE|FAILURE_ROUTE|ANY_ROUTE}, {"async_task_route", (cmd_function)w_async_task_route, 1, fixup_async_task_route, 0, REQUEST_ROUTE|FAILURE_ROUTE|ANY_ROUTE}, {0, 0, 0, 0, 0, 0} }; Regards, RajaGopal Maddi

2 1

Hello
by Subba Krishna MG 29 Sep '17

29 Sep '17

Hello, i am using Kamailio for SIP Presence server. When ever the Presence PUBLISH is received on kamailio, kamailio is sending 200 OK with out Expires IE. I want Kamailio to send Expires IE with 200 OK. How do i modify it. Can you please help me in this regard. Thank u MGSK

1 0

Retrieve remote IP and port
by Iskren Hadzhinedev 29 Sep '17

29 Sep '17

Hi list, How can I reliably get the sender’s IP address? |$si| and |$sp| are returning the server IP and Port. I also tried using |$Ri| and |$Rp| but it yields the same results. Inspecting the packet shows the sender’s true IP:Port pair in the |Via:| header, but the |From:| and |To:| contain the kamailio server’s public IP address. Kind regards, -- /Iskren Hadzhinedev/

2 3

Perl module missing in kamailio 5.0.3
by Ginhoux, Patrick 29 Sep '17

29 Sep '17

Hi, I'm thinking to upgrade my kamailio 5.0.1 running on a Rhel 7.1 server to 5.0.3 . Looking at the repository for Rhel7 to download the rpm, I don't find the perl rpm. For 5.0.1, it is kamailio-perl-5.0.1-1.1.x86_64.rpm. Can you help to get it for 5.0.3 ? Thanks in avance. Cordialement Patrick GINHOUX

2 4

how to debug kamailio c code
by Sergey Safarov 28 Sep '17

28 Sep '17

I have issue with kazoo module call sprintf <https://github.com/kamailio/kamailio/blob/master/src/modules/kazoo/kz_pua.c…> not work properly. body variable contains values like \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 Must be XML string <?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" entity="2496"> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t8No1hQnp65U33jS1oqFMgxbIRPq-iFP"><status><basic>open</basic></status></tuple><note xmlns="urn:ietf:params:xml:ns:pidf">On the Phone</note><dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="1"><rpid:activities><rpid:on-the-phone/></rpid:activities><dm:note>On the Phone</dm:note></dm:person></presence> But i cannot find/understand where is must be executed conversion hex2string I want use gdb inspect memory vars but i cannot. Please advice how to do it.

1 0

Re: [SR-Users] Question about Kamailio Performance
by Jiankai Zhu 28 Sep '17

28 Sep '17

Thank you guys for your quick response. I am new to Kamailio so all your information are very helpful for me. My setting: children=8 in kaimailio.cfg . I did not change it and leave it by default on my 4 cores system. My thought: I think it may related to Mysql since cpu and iowait is low and the problem happens only when I add more subscribers to mysql database. I will try according to : 1: Dmitri combination of mysql and htable as mysql cache 2: Mathias tune MySQL server 3: I think ram database should help on this. But it seems these is little document on redis. I am looking into SQLite3 memory mode but if there is any related reference / link will be appreciated. Thanks with regards, David David 2017年9月28日下午02:59，Daniel-Constantin Mierla <miconda(a)gmail.com> 写道： Hello, it seems that interaction with mysql is a bottleneck. What is the value of children parameter in kamailio.cfg? Cheers, Daniel On 28.09.17 04:00, Jiankai Zhu wrote: My goal: make sure when using kamailio as engine I can achieve: on systems with 4GB memory, Kamailio can serve a population over 300 000 online subscribers <from: https://www.kamailio.org/w/features/ > Test bed: CPU: 4 cores , 4G RAM Kamailio 5 with Mysql. Test source：another server running SIPp 10k subscriber in Mysql . 6000 Register Per Second , reuse 10k user to register , it works fine. If register 50 every second for per user it seem 300 k subscribers should work. (about 300% cpu used of totally 400% cpu. All requests finished without retransmit and fail. ) add 90k subscriber into Mysql . 100k subscirbers in Mysql . 1000 Register Per Second , a lot of failure and can not work . (cpu are not used much at that time. IO is not high.) My question : What is the root cause of bad performance when there is 100k subscribers' data in the Mysql database. How can I deploy a kamailio with 300k subscriber in database, and support 6000 register per second at the same time ? Maybe use ram database instead of Mysql? Thanks with regards, David _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 1

mysql installation error
by Timothy oladapo olawuyi 28 Sep '17

28 Sep '17

Dear All, Please help me with error messages. Am trying to configure kamailio MYSQL as DBEGINE base on the tutorial available in the SIP ROUTING WITH KAMAILIO book, when I executed the /usr/local/sbin/kamdbctl create and got the Error messages below. ERROR: could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' Here this the version of kamailio I installed. ./kamailio -I. Print out of kamailio internals Version: kamailio 4.4.6 (x86_64/linux) 75f13d Default config: /usr/local/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_LISTEN=16 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 75f13d Compiled with: gcc 4.9.2 Compiled on: 10:56:51 Sep 28 2017 Thank you for flying kamailio! Regards Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 |Skype: daptims | <htttp://www.pethahiah.com/> htttp://www.pethahiah.com

3 9

How to do replace_body_all with Kemi python version kamailio 5.0.x
by Muhammad Zaka 28 Sep '17

28 Sep '17

Hi We are experiencing issue with python script KSR. textops.replace_body_all using branch 5.0.2: - Example Code: KSR.textops.replace_body_all("a=sendonly", "a=sendrecv") It looks like that the module textops and textopsx are not available. Using $rb or $msg(body) pseudo variables isn't working as they are readonly. Is there any other way to replace the body? Many Thanks Regards Muhammad Zaka

2 3

CentOS 7 TLS Unable to set the memory allocation functions
by Ryan Wagoner 28 Sep '17

28 Sep '17

I had this issue with CentOS 7.3 and chalked it up to the openssl 1.0.1e bug mentioned on the list https://lists.kamailio.org/pipermail/sr-users/2013-August/079381.html However I just updated to CentOS 7.4 with openssl 1.0.2k-8 and the issue still exists with Kamailio 4.4.6-2.2. I've tried moving the loadmodule "tls.so" to the top as well. ERROR: tls [tls_init.c:526]: tls_pre_init(): Unable to set the memory allocation functions ERROR: tls [tls_init.c:528]: tls_pre_init(): libssl current mem functions - m: 0x7fc56d19c0c0 r: 0x7fc56d19c5a0 f: 0x7fc56d19c4c0 ERROR: tls [tls_init.c:530]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) Does anybody have TLS working with RHEL 7 or CentOS 7? Thanks, Ryan

2 2

Question about Kamailio Performance
by Jiankai Zhu 28 Sep '17

28 Sep '17

My goal: make sure when using kamailio as engine I can achieve: on systems with 4GB memory, Kamailio can serve a population over 300 000 online subscribers <from: https://www.kamailio.org/w/features/ > Test bed: CPU: 4 cores , 4G RAM Kamailio 5 with Mysql. Test source：another server running SIPp 10k subscriber in Mysql . 6000 Register Per Second , reuse 10k user to register , it works fine. If register 50 every second for per user it seem 300 k subscribers should work. (about 300% cpu used of totally 400% cpu. All requests finished without retransmit and fail. ) add 90k subscriber into Mysql . 100k subscirbers in Mysql . 1000 Register Per Second , a lot of failure and can not work . (cpu are not used much at that time. IO is not high.) My question : What is the root cause of bad performance when there is 100k subscribers' data in the Mysql database. How can I deploy a kamailio with 300k subscriber in database, and support 6000 register per second at the same time ? Maybe use ram database instead of Mysql? Thanks with regards, David

3 2

Subscribe Reinvites totag
by Soltanici Ilie 27 Sep '17

27 Sep '17

Hi All, I have system setup where there are few Asterisk servers behind Kamailio. Everything is working properly, the issue only is with the Incoming Subscribe's with to-tag from endpoints. Below you can see the invite: ---------------------------------------------------------------------------------------------------------- 2017/09/27 19:40:29.995598 192.168.2.33:5060 -> 10.25.23.1:5060 SUBSCRIBE sip:10000211@10.25.23.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.33:5060;branch=z9hG4bK-fc65f67d From: "Cisco" <sip:211@10.25.23.1>;tag=258a856c69c288 To: <sip:10000211@10.25.23.1>; tag=02cca2b6-1d9b-4edc-b1de-fdd78f8f9ab0 Call-ID: 4ce04a70-6128897b(a)192.168.2.33 CSeq: 1003 SUBSCRIBE Max-Forwards: 70 Route: <sip:10.25.23.1;lr;r2=on;ftag=258a856c69c288;nat=yes>, <sip:10.1.1.101;lr;r2=on;ftag=258a856c69c288;nat=yes> Proxy-Authorization: Digest username="211",realm="10.25.23.1",nonce="Wcvq0VnL6aUbQ3c+nZa5xbh8KMVoOwAq",uri="sip:10000211@10.25.23.1",algorithm=MD5,response="155f68d66b3c6ad035f7db1ec5732020" Contact: "Cisco" <sip:211@192.168.2.33:5060> Accept: application/dialog-info+xml Expires: 1800 Event: dialog User-Agent: Cisco/SPA504G-7.6.2 Content-Length: 0 ---------------------------------------------------------------------------------------------------------- Receiving such Invite Kamailio are routing to RELAY route (from WITHINDLG) and showing such error: ERROR: rr [loose.c:107]: find_first_route(): failed to parse Route HF It seems that Kamailio cannot find such Dialog, but why is happening? where is it destroyed? How can i solve this? Route (WITHINDLG) is as default, below you can see it as well. ----------------------------------------------------------------------------------------------------------- route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if ( is_method("ACK") ) { route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); } exit; } ----------------------------------------------------------------------------------------------------------- Thank You.

1 0

Next IRC devel meeting
by Daniel-Constantin Mierla 27 Sep '17

27 Sep '17

Hello, to sync properly for the next major release (v5.1.0) and ongoing development, I propose an IRC devel meeting for next week, on Wednesday, Sep 27, 2017. An alternative would be the following day, Sep 28, or if there are many devs that want to attend and cannot do it these days, we can look at another date. Just propose new day and time. I created a wiki page to collect the topics that are wanted to be discussed: - https://www.kamailio.org/wiki/devel/irc-meetings/2017b Fell free to add there or reply to the mailing list with what you think it is relevant to discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 3

dialog:active_dialogs counter is steadily increasing
by Andrew Pogrebennyk 27 Sep '17

27 Sep '17

Hi, using kamailio 4.4.5, we have noticed that the active_dialogs counter is not accurate and there seems to be some kind of "leak" present at least on higher loaded systems. For example, here active_dialogs was reporting just a bit over 1000 dialogs, but the numbers of dialogs confirmed and not just do not sum up together to that value: # grep -c 'state:: 1' dialogs_20170919_1712.txt 5 # grep -c 'state:: 2' dialogs_20170919_1712.txt 62 # grep -c 'state:: 3' dialogs_20170919_1712.txt 0 # grep -c 'state:: 4' dialogs_20170919_1712.txt 531 # grep -c 'state:: 5' dialogs_20170919_1712.txt 27 What could be the possible reasons or issues that lead to that? We are thinking now about switching statistics collection to: kamctl fifo dlg_list | grep -c 'state:: 4' or reading the count directly from the database, as it seems to only write the dialogs with state 4 to mysql and those numbers seem OKish. Any ideas? Regards, Andrew

3 6

How to do msg_apply_changes with Kemi python version 5.0.x
by Muhammad Zaka 27 Sep '17

27 Sep '17

Hi We are experiencing issue with python script KSR. textopsx.msg_apply_changes using branch 5.0.2: - Example Code: KSR.textopsx.msg_apply_changes() Error: Sep 22 15:25:19 sbc99-kam-dev /usr/sbin/kamailio[23141]: PYT {INVITE}: ERROR: app_python [python_support.c:142]: python_handle_exception(): python_exec2: Unhandled exception in the Python code:#012Traceback (most recent call last):#012 File "/etc/kamailio/kamailio-kemi-python.py", line 64, in ksr_request_route#012 if self.ksr_route_reqinit(msg)==-255 :#012 File "/etc/kamailio/kamailio-kemi-python.py", line 173, in ksr_route_reqinit#012 KSR.textopsx.msg_apply_changes()#012AttributeError: 'module' object has no attribute 'textopsx' Is anybody have any idea how to workaround and apply message changes in python? Many Thanks Regards Muhammad Zaka

2 1

topos and record-route headers
by Giuseppe Pandolfi 27 Sep '17

27 Sep '17

Hi, I'm trying to use topos module but I have a problem with record-route header in re-invite for on hold. Uac is a subscriber of sipwise platfrom (vm open source version); my kamailio version is: /# kamailio -v// //version: kamailio 5.1.0-dev7 (ppc/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/ and uas is a subscriber of kamailio. The call flow is: UAC Sipwise Kamailio UAS | | (topos active) | |-------- Invite -------->|---------- (1)Invite ----->|------- (2)Invite ----->| |<------- 180/200 ------|<--- (4)180/(4)200 ----|<------- 180/200 ------| |---------- Ack --------->|----------- (5)Ack ------>|-------- (6)Ack ------->| |--- Invite (hold) ----->|-- (7)ReInvite(hold)-->|-- (8)Invite (hold)--->| |<--------- 200 ----------|<--------- (9)200 -------|<--------- 200 ----------| | corrupt | corrupt | | |---- No Ack Sent-------|---- No Ack Sent-------|---- No Ack Sent-------| *The problem:* the 200 ok of re-invite is corrupt by topos, it hasn't via header and contact isn't modified, then the uac can't send the ack. In the kamailio trace I see the ERROR below: *E!: U 22/09/2017 10:24:19:880 proxy0: ERROR: topos [tps_storage.c:1254]: tps_db_update_dialog(): no valid dlg uuid* *The questions are:* Can I use topos for this scenario? Is it possible that topos have bug when receive a re-invite with record-route headers? Infact, in simple scenario without record-route (call and hold between tow subscriber) everything works! *My opinion is*: when topos receive a request with record-route generates from record-route the new aleg and bleg tags (atpsh-xxxxx, btpsh-yyyyyy) but don't save their in internal database if the request is internal at dialog. Then a reintive whit record-route headers, generate new aleg/bleg tags but they aren't in database, then the response 200 ok it isn't elaborate. below messages details, attached kamailio log and pcap trace. tanks in advance for support, Giuseppe* * * * *(1)Invite:* INVITE sip:100@192.168.110.79:5060;transport=udp SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79> CSeq: 10 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri P-Asserted-Identity: <sip:390735594284@192.168.110.132> Content-Type: application/sdp Content-Length: 276 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> [sdp] *(2)Invite:* INVITE sip:100@192.168.1.73:5080;line=4d3148ee1d889aa SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK1a43.c9b996baf80be68d288d2de06cd9b212.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79> CSeq: 10 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri P-Asserted-Identity: <sip:390735594284@192.168.110.132> Content-Type: application/sdp Content-Length: 276 Contact: <sip:btpsh-59c55b4a-4fa6-1@192.168.110.79> [sdp] *(3) 180 Ringing* SIP/2.0 180 Ringing From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 10 INVITE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0,SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 Contact: <sip:atpsh-59c55b4a-4fa6-1@192.168.110.79> Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes>,<sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> *(4) 200 OK* SIP/2.0 200 OK From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 10 INVITE Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 181 Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0,SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 Contact: <sip:atpsh-59c55b4a-4fa6-1@192.168.110.79> Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes>,<sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> [sdp] *(5) ACK* ACK sip:atpsh-59c55b4a-4fa6-1@192.168.110.79 SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.4446035c867218358dbd848f773b5d09.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKspF2zas~;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 10 ACK Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Content-Length: 0 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> *(6) ACK* ACK sip:100@192.168.1.73:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK1a43.b1cce71dc1b004e442faca53cd6d8994.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 10 ACK Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Content-Length: 0 Contact: <sip:atpsh-59c55b4a-4fa6-2@127.0.0.1> *(7) Re-Invite * INVITE sip:atpsh-59c55b4a-4fa6-1@192.168.110.79 SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK2a43.3723a1a3f343f1f1974a66f4a62f41a9.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKf6peiap7;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 11 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Content-Length: 276 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> [sdp] *(8) Re-Invite* INVITE sip:100@192.168.1.73:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK2a43.a4e5f5baf85414d4471a1e817bdc1e00.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 11 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Content-Length: 276 Contact: <sip:atpsh-59c55b4a-4fa6-3@127.0.0.1> [sdp] * **(9) 200 OK* SIP/2.0 200 OK From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 11 INVITE Contact: <sip:100@192.168.1.73:5080> Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 211 [sdp] ------------------------------------------------------------------------

2 5

UACREG - answering to 401 on invite
by Jean Cérien 26 Sep '17

26 Sep '17

Hello I'm using Kamailio 5.0.1 With the UACREG module, I am registering to a remote provider. Register goes out, 401 back, Register goes out with nonce & co, OK Later, when I send an invite, the provider issues an 401 Unauthorized. I guess it expects me to resubmit an INVITE with the authentication data, but I dont see how to do that. Any help would be appreciated ! J.

2 6

How to change SDP in 200 Ok reply
by Jurijs Ivolga 26 Sep '17

26 Sep '17

Hi, I have difficult times to edit SDP in 200 ok replies, I need to change RTP/SAVPF to UDP/TLS/RTP/SAVPF in all 200 Ok. Here is code snippet: onreply_route { if ( $rm == "INVITE" && status=="200") { if (search_body("a=fingerprint")) { if (search_body(" RTP/SAVPF ")) { xlog("L_INFO","200 OK DTLS call.\n"); replace_body_all(" RTP/SAVPF "," UDP/TLS/RTP/SAVPF "); } } } } What I'm doing wrong? I think my code is executed against SDP of Invite, but not 200 OK. Jurijs

2 2

Best variant to share variables between two dialogs?
by Kristijan Vrban 26 Sep '17

26 Sep '17

Hi, based on Daniels example from: http://blog.miconda.eu/2010/01/best-of-new-in-kamailio-300-19-htable.html ... modparam("htable", "htable", "acalls=>size=8;") ... route { ... if(is_method("INVITE") && !has_totag()) { # a new call if($shtcv(ht=>^$fU$)>=3) { send_reply("403", "limit exceeded"); exit; } $sht(acalls=>$ci) = $fU; t_on_failure("NEW_INVITE"); } if(is_method("BYE")) { $sht(acalls=>$ci) = $null; } ... } failure_route[NEW_INVITE] { $sht(acalls=>$ci) = $null; } I use this htable variant to share data between two dialogs (store Contact from one dialog, and re-write other R-URI with it). Referenced by call-id: ... $avp(referenceID) = $ci + "-" + $avp(direction); $sht(a=>$avp(referenceID)) = $ct; ... ... $avp(referenceID) = $ci + "-" + $avp(direction); avp(newru) = $sht(a=>$avp(referenceID)); avp_subst("$avp(newru)", "/(<)(.*)(>)/\2/"); xlog("L_NOTICE", ">>>>>>>> ReWrite ru with:$avp(newru)\n"); $ru = $avp(newru); ... Is this the best method to share data between two dialogs? Because you have to take care yourself with failure_route and so on, to free (set back to $null) them. Or is there a more elegant way for that task? Best Kristijan

1 0

Send 407 for Invite
by Logeshwaran G 26 Sep '17

26 Sep '17

Hi, How to send the 407 proxy authentication for Invite message? Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 3

Memory allocation issue
by Logeshwaran G 26 Sep '17

26 Sep '17

Hi, I am having the below Error while executing the kamailio cfg file. Before an hour it executed successfully, There is no changes in cfg. loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(27575) : <core> [core/mem/shm.c:105]: shm_getmem(): could not attach shared memory segment: Cannot allocate memory 0(27575) : <core> [core/mem/q_malloc.c:1203]: qm_malloc_init_shm_manager(): could not initialize qm shm memory pool Too much qm shm memory demanded: 67108864 bytes 0(27575) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/config/kamailio_final.cfg, line 302, column 39: Can't initialize shared memory ERROR: bad config file (1 errors) Input please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

1 0

Docker image
by Sergey Safarov 25 Sep '17

25 Sep '17

Hello guys Yes i packaged kamailio to docker container <https://hub.docker.com/r/safarov/kamailio3/>. This container is ready to run kamailio in docker environment. Could you make review of this. Dockerfiles you can review at https://github.com/kamailio/kamailio/tree/master/pkg/kamailio/alpine_docker Build process logs you can review at https://hub.docker.com/r/safarov/kamailio3/builds/bl9rv6ygeubwnydh4an2tuc/ Sergey

3 3

Astricon 2017
by Daniel-Constantin Mierla 25 Sep '17

25 Sep '17

Hello, Kamailio will be present at Astricon 2017 having a stand in the expo area (Orlando, FL, during October 3-5 -- more details at www.astricon.net). Should you participate at the event and have flyers/other materials with products/solutions/services that use Kamailio or giveaways, you are welcome to join the booth or just drop them there, we are more than happy to promote and spread them. Also, you are very welcome to just stop by and have a chat. Besides the activity in the exhibition area, I and Fred Posner will have presentations about using Kamailio together with Asterisk or other VoIP applications. Be sure you do not miss them, the agenda is available at: - http://www.asterisk.org/community/astricon-user-conference/schedule Expect a consistent group from Kamailio community, it's a great event to be at! Looking forward to meeting many of you next week in Orlando! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 3

Re: [SR-Users] [sr-dev] SIP ping, sockets and multi homed
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, After deep investigation I found out that the issue is happening because of the mechanism Kamailio uses to find the most appropriate socket combined with the virtual IP usage.... For socket selection, Kamailio first asks Linux which would be the routing for a certain destination, then matches this outcome with the list of listened sockets. By default Linux will always return the local IP instead of the virtual one. So if Kamailio only binds to the virtual IP, the Linux's returned IP won't match our socket list and then the socket selection fails.... So currently Kamailio active/stand-by HA setup with virtual IP in a multi-homed environment does not work properly... We can overcome this by listening to all addresses, but then Kamailio will decide to use the local IP instead of the virtual one, and then the HA idea end up broken.... We can also force socket for everything. For regular call routing, this options actually works fine if the script always force some socket, but it seems to be not always possible for local generated messages. By disabling mhomed and using the dispatcher socket attr to send the OPTIONS message it works for example. But for the LCR module we have no option because apparently the LCR module has only one socket option, and so I am not able have a different socket per gateway. We can also explicitly set the src address for a routing rule to be the virtual IP instead of the local one. This makes Linux to return the virtual IP instead of the local one. This solution seems fine, but since it is done at linux level, every other service on the box would be affected and every route on the box would needed to be changed.. Another problem I found out is that I am not able to force the socket for local generated messages. It actually crashes Kamailio, and I always get a "core was not generated" msg.... anyone aware of that? The actual fix for this, would be having Kamailio to search the most appropriate socket in a different manner. It could loop the sockets and see which one reaches the destination instead of asking Linux to do it, or it could have some routing directives which could be consulted before selecting a socket.... (just ideas, not sure how feasible they are...) I am considering using the Linux routing change option, but has anyone faced such situation before? Regards, Patrick Wakano On 24 August 2017 at 09:18, Patrick Wakano <pwakano(a)gmail.com> wrote: > Hello Daniel, > I still couldn't make this SIP probing work.... So currently my setup > cannot actively check for GW availability... > Any idea about what may be causing it? > > Thanks for your time, > Patrick Wakano > > On 16 August 2017 at 13:01, Patrick Wakano <pwakano(a)gmail.com> wrote: > >> Forgot to mention it is a centos7 box... >> >> On 16 Aug. 2017 11:02 am, "Patrick Wakano" <pwakano(a)gmail.com> wrote: >> >>> Hi Daniel! >>> Very grateful for your attention! >>> >>> Apologies for lacking important info, check below the version and >>> sockets. >>> It may be worth mentioning these IPs are virtual ones and I am using net.ipv4.ip_nonlocal_bind >>> = 1(in this case the virtual IP is assigned to the box) >>> Interesting that it has a "mhomed: no" for all sockets but I do have >>> mhomed=1 in my cfg file.... could that be the issue? >>> Also there is no message before this one: /usr/sbin/kamailio[6164]: >>> ERROR: <core> [core/forward.c:181]: get_out_socket(): no socket found >>> This is the first one that pops when dispatcher tries to ping.... >>> >>> Thanks, >>> Patrick >>> >>> kamailio -v >>> version: kamailio 5.0.2 (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 12:03:49 Jun 26 2017 with gcc 4.8.5 >>> >>> kamcmd> core.sockets_list >>> { >>> socket: { >>> proto: udp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: udp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 8080 >>> mcast: no >>> mhomed: no >>> } >>> } >>> >>> >>> On 15 August 2017 at 17:24, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> what are the sockets kamailio is listening on? See `kamcmd help` for >>>> the command that lists the sockets. >>>> >>>> Also, what is the log messages before the first one you pasted, there >>>> should be another one printed from get_out_socket(). >>>> It is always important to provide kamailio version (kamailio -v) and >>>> the operating system you are using. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 14.08.17 14:41, Patrick Wakano wrote: >>>> >>>> Hello Kamailio list, >>>> >>>> Hope you all doing well! >>>> >>>> I am stuck with one problem that should be incredibly easy to be done, >>>> but apparently it is not.... I want my dispatcher, LCR and nathelper >>>> modules to always SIP ping my SIP clients. I have a Kamailio based SBC that >>>> is separating two networks, and so I am using the 'mhomed' flag, but none >>>> of these SIP pings are working.... for now I will only ask about the >>>> dispatcher.... >>>> My dispatcher module has to talk with SIP servers in both networks, the >>>> thing is when I enable SIP ping for it (modparam("dispatcher", >>>> "ds_probing_mode", 1)), it gives me this type of error: >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[6164]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> After research it seems I need to manually set a "socket" attribute for >>>> each of my GWs in dispatcher list. After putting the socket parameter for >>>> the GW, I got another error: >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/socket_info.c:2046]: >>>> parse_protohostport(): bad port number in udp:192.168.33.100:5060 >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[14190]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> What can be wrong in my port info in the socket config? (I tried the >>>> socket with and without quotes but had no luck...) >>>> select * from dispatcher where id=2; >>>> id | setid | destination | flags | priority >>>> | attrs | description >>>> ----+-------+-------------------------+-------+----------+-- >>>> -------------------------------------------+--------------------- >>>> 2 | 2 | sip:192.168.33.110:6060 | 0 | 0 | socket="udp: >>>> 192.168.33.100:5060";rweight=50 | >>>> (1 row) >>>> >>>> Anyhow, I was expecting the 'mhomed' parameter to do the socket >>>> selection for me. Regular call routing either by the dispatcher or by LCR, >>>> also seems to require me to explicitly set the socket otherwise Kamailio >>>> (or the OS) may end up using the wrong socket. So, another question is why >>>> 'mhomed' does not dictate the whole socket selection Kamailio has to do >>>> regardless the module it is using to send SIP requests? >>>> >>>> Sorry for the long e-mail.... >>>> Best regards, >>>> Patrick Wakano >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Development Mailing Listsr-dev@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Kamailio Advanced Training - www.asipto.com >>>> Kamailio World Conference - www.kamailioworld.com >>>> >>>> >>> >

2 2

LCR Ping GWs
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, I was trying to activate continuous SIP ping from the LCR module, which apparently was available in the past, but currently it is not possible anymore. As I could check It can only ping inactive GWs. Question is, why such change was promoted? We have to send a call to the GW to detect it has died instead of actively monitoring them and so avoid this useless call attempt. It is not that big deal, but it does gives me a better monitoring of my providers (specially the ones not used so frequently) health. Regards, Patrick

1 0

Enabling Redirection in Kamailio SIP Server
by Anas Rasheed 23 Sep '17

23 Sep '17

How can we enable redirection in Kamilio SIP Server using Siremis User Interface for it? Can anyone provide us with the tutorial of it?

1 0

Memory configuration
by Joe Baran 23 Sep '17

23 Sep '17

I feel fairly confident I’m running into a memory tuning issue, but I’m struggling to determine the next step. Can someone give me some guidance on how to adjust settings to allow the system to stop crashing kamailio? Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7fa9eeeea000, 144); Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fa9eeeea000, 144) called from kazoo: kz_amqp.c: kz_amqp_async_query(1357), module: kazoo; Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: kazoo [kz_amqp.c:1359]: kz_amqp_async_query(): failed to allocate kz_amqp_cmd in process 29468 Sep 14 18:25:15 server1 kamailio[29468]: INFO: <script>: kGddp3V0bA6nitnk25dB4w..|log|failed to send Kazoo query for authentication xxxxxxxxxxxxxxxxxx Sep 14 18:25:15 server1 kamailio[29468]: ERROR: tm [t_reply.c:494]: _reply_light(): cannot allocate shmem buffer Sep 14 18:25:15 server1 kamailio[29468]: ERROR: sl [sl.c:271]: send_reply(): failed to reply stateful (tm) Sep 14 18:25:16 server1 kamailio[29483]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 108 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:743]: handle_sigs(): child process 29438 exited by a signal 11 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Thanks, Joe Baran

3 6

Shared htable between servers
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi list, Is it possible to have a htable shared between servers on different machines? I have two kamailio proxies on different servers for resilience and I'm storing call information in a hash table. Then I can route calls for the same user to the respective PBX. The reason for this is to have attended transfers working reliably (attended transfer between PBXes are a mess). Thank you! Kind regards, -- /Iskren Hadzhinedev/

2 1

Call transfer between pbx servers behind kamailio proxy
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi everyone, I'm having a hard time transferring calls when the users are on different FreeSWITCH servers behind kamailio. The users are distributed as follows: user1(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user2(a)sip.test.com-> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user3(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH2 When user1 calls user2 and user2 does an attended transfer to user3, the following happens: 285 67.220453 USER2_IP -> KAMAILIO_IP SIP 726 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 286 67.220674 KAMAILIO_IP -> FreeSWITCH1_IP SIP 877 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 287 67.223651 FreeSWITCH1_IP -> KAMAILIO_IP SIP 925 Status: 202 Accepted | 288 67.223753 KAMAILIO_IP -> USER2_IP SIP 838 Status: 202 Accepted | 289 67.225502 FreeSWITCH1_IP -> KAMAILIO_IP SIP/SDP 1225 Request: INVITE sip:user3@sip.test.com | 290 67.387323 KAMAILIO_IP -> 69.172.200.109 SIP/SDP 1420 Request: INVITE sip:user3@sip.test.com | Turns out 69.172.200.109 is the actual IP address of sip.test.com (I've probably should have gone with sip.example.com or something else). What is the proper way to route that INVITE back to the other FreeSWITCH server so that the transfer would work? Thanks! Kind regards, -- /Iskren Hadzhinedev/

2 2

how to play ring tune when callee decline s
by 赵国杰 22 Sep '17

22 Sep '17

Hi guys, I want the caller to play a short audio(like "the number your are calling is busy") when the callee declines the call. How can i do that? Best Regards Jesse

4 21

Typos in Kamctl Stats results
by Noah Mehl 21 Sep '17

21 Sep '17

Hey, I’m noticing that the results from the websockets module when running # kamctl stats there are some odd results that may just be typos? "websocket:ws_msrp_max_concurrent_connections = 0", "websocket:ws_msrp_max_concurrent_connectons = 0”, "websocket:ws_sip_max_concurrent_connections = 0", "websocket:ws_sip_max_concurrent_connectons = 0", Thanks! ~Noah

3 2

Encryption of voice stream
by Vasan VS 21 Sep '17

21 Sep '17

Hello all, Kindly let us know if we can have a mechanism to have the voice stream in using the SIP server completely encrypted as well as prove that it is encrypted using any speific mechanism. Thanks in advance Vasan

3 2

COALESCE Transformation?
by Brandon Armstead 20 Sep '17

20 Sep '17

Hello all, I've checked the docs and did some brief searching, curious if this transformation or method exists currently? Such as: https://dev.mysql.com/doc/refman/5.7/en/comparison-operators.html#function_… Thanks! -- Sincerely, Brandon Armstead CTO / CRYY.com

1 0

private memory leak - but where?
by Sebastian Damm 20 Sep '17

20 Sep '17

Hi, I'm trying to debug a Kamailio which runs out of memory after a while. But following the hints at https://www.kamailio.org/wiki/tutorials/troubleshooting/memory doesn't help. I know it is the private memory that I'm missing. I already increased the pkg size to 16M, but somehow that seems to be not enough. The Kamailio receives HEP messages, sends them to a lua script where it is processed. Overall there are not more than 10 custom private variables used. In the log I'm seeing these messages: Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: app_lua [app_lua_api.c:726]: app_lua_run_ex(): error executing: handle_packet (err: 2) Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <script>: SCRIPT: failed to execute lua function! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7f9744eea010, 232); Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f9744eea010, 232) called from core: core/parser/msg_parser.c: get_ hdr_field(116), module: core; Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:118]: get_hdr_field(): out of memory Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:330]: parse_headers(): bad header field [Via: SIP/2.0/WSS o6m] Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: pv [pv_core.c:647]: pv_get_callid(): cannot parse Call-Id header kamcmd pkg.stats for this process gives me this output. { entry: 61 pid: 19134 rank: 61 used: 8748080 free: 4384 real_used: 16772832 total_size: 16777216 total_frags: 30 } Now I tried to get a dump of what is in use, following the guidelines on the wiki page mentioned above, but without success. I can easily start and stop the traffic coming in to Kamailio (by starting or stopping the captagent sending it there), but I can't get any output. Here's what I did: captagent stop kamcmd cfg.set_now_int core memlog 1 kamcmd cfg.set_now_int core mem_dump_pkg 19134 captagent start ; sleep 1 ; captagent stop (I started captagent so that the process actually had something to do because that's what the wiki page says it needs for printing memory information.) During that second, there were quite a lot of packets running through this process, but I did not get a memory dump. A shm_summary dump however works as expected. What am I missing? How do I get more information about what this process is using its memory for? TIA Sebastian

2 6

sipp audio file
by Diego Nadares 19 Sep '17

19 Sep '17

Hi guys, Any body know how to create pcap (with audio) file to use with sipp?  <nop> <action> <exec play_pcap_audio="../scenarios/pcap/demo.cap"/> </action> </nop> I tried with wireshark capturing rtp but I'm doing something wrong i think. Thanks! Diego

4 4

jsonrpcc problem
by Adam Kuśmirek 19 Sep '17

19 Sep '17

Hello, I have a problem with jsonrpcc module. OS is Debian Jessie. I tried with packeged kamailio 4.4 and compiled kamailio 5.0. Kamailio starts properly. Routing block is very simple: route { jsonrpc_request("get_source", "{'ipv4': '$si', 'port': $sp}", "RESPONSE", "ERROR", "$var(result)"); exit; } route[RESPONSE] { xlog("Result received: $var(result)"); } route[ERROR] { xlog("Error received: $var(result)"); } If any message arrives, kamailio sends it to my rpc_server (netstring encoded): 106:{'id': 2, 'jsonrpc': '2.0', 'method': 'get_source', 'params': {'ipv4': '10.20.121.231', 'port': 5060}}, My server decodes the message and immediately answers with netstring encoded response: 47:{"jsonrpc": "2.0", "result": "200 OK", "id": 2}, At this moment kamailio logs errors and disconnects from rpc server: root@kamailio-test:/etc/kamailio# 4(925) DEBUG: <core> [parser/msg_parser.c:595]: parse_msg(): SIP Request: 4(925) DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:kamailio.conpeek.com> 4(925) DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> 4(925) DEBUG: <core> [parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1033201528>; state=16 4(925) DEBUG: <core> [parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 4(925) DEBUG: <core> [parser/msg_parser.c:486]: parse_headers(): Via found, flags=2 4(925) DEBUG: <core> [parser/msg_parser.c:488]: parse_headers(): this is the first via 4(925) DEBUG: <core> [receive.c:178]: receive_msg(): After parse_msg... 4(925) DEBUG: <core> [receive.c:221]: receive_msg(): preparing to run routing scripts... 4(925) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 4(925) DEBUG: <core> [parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 4(925) DEBUG: <core> [parser/msg_parser.c:171]: get_hdr_field(): <To> [40]; uri=[sip:1000@kamailio.conpeek.com] 4(925) DEBUG: <core> [parser/msg_parser.c:172]: get_hdr_field(): to body ["1000" <sip:1000@kamailio.conpeek.com> ] 4(925) DEBUG: <core> [parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:183]: get_hdr_field(): content_length=0 4(925) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 4(925) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=43020, isACK=0 4(925) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(925) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 4(925) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: 920bfe28c581e1bf18554e3a9b234ceb 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 4(925) DEBUG: <core> [receive.c:318]: receive_msg(): cleaning up *0(921) ALERT: <core> [main.c:740]: handle_sigs(): child process 929 exited by a signal 11* * 0(921) ALERT: <core> [main.c:743]: handle_sigs(): core was generated* 0(921) INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD 0(921) DEBUG: <core> [main.c:757]: handle_sigs(): terminating due to SIGCHLD 7(928) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 6(927) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 5(926) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 4(925) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 3(924) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 2(923) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 1(922) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 0(921) DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start 0(921) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table 0(921) DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fb99a822350 from h_table.c:449 0(921) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores 0(921) DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists 0(921) DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done 0(921) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 0(921) DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock 0(921) DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc 0(921) DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc The problem occures also if rpc server does not reply, i think when trying to start ERROR route. If I try with jsonrpc_notification, everything is OK. Any help ? Regards Adam

2 2

See how many workers are actually processing packets?
by Sebastian Damm 19 Sep '17

19 Sep '17

Hi, I think this has been answered before, but I don't know the answer. Is there a way to find out, how many workers are busy right now processing packets? For a new setup I don't know whether I need 16 or 96 UDP listeners, but I'd like to scale it correctly. The setup just inspects HEP messages and sends some of them to a RabbitMQ. So TM or SL stats probably won't help. Best Regards Sebastian

1 0

Proxy Invite to registered natted user
by eskornev 19 Sep '17

19 Sep '17

Hi All Trying to set Kamailio as proxy in front of Asterisk. When natted user registers is remembered in Kamailio via save("location"). Kamailio forwards Register to * with new contact field (pointing to Kamailio). But when server originates Invite to user, kamailio transfer request to "fake" address that was in original Register contact field (getting it from lookup("location")). How can i set Kamailio to send Invite to source ip and port of user?

2 1

kamailio jsonrpc client
by Aidar Kamalov 19 Sep '17

19 Sep '17

Hello, cannot get working rpc client n kamailio. [root@sipchel kamailio]# grep 'json\|jansson' kamailio-default.cfg loadmodule "jsonrpcs.so" loadmodule "jansson.so" loadmodule "janssonrpcc.so" # jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 3) # jansson modparam("janssonrpcc", "server", "conn=user_db;addr=127.0.0.1;port=5060;priority=10;weight=10"); .... request_route { ... janssonrpc_request("user_db", "core.version", '', "route=RESPONSE;retry=1"); ... } .... event_route[xhttp:request] { if ($hu =~ "^/jsonrpc") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } With this config i got error Sep 19 08:52:15 sipchel /usr/sbin/kamailio[39891]: INFO: janssonrpcc [janssonrpc_connect.c:294]: bev_connect(): Connecting to server 127.0.0.1:5060 for conn user_db. Sep 19 08:52:15 sipchel /usr/sbin/kamailio[39891]: INFO: janssonrpcc [janssonrpc_connect.c:277]: bev_connect_cb(): Connected to host 127.0.0.1:5060 ... Sep 19 08:52:20 sipchel /usr/sbin/kamailio[39891]: WARNING: janssonrpcc [janssonrpc_request.c:242]: schedule_retry(): Number of retries exceeded. Failing request. Sep 19 08:52:20 sipchel /usr/sbin/kamailio[39891]: WARNING: janssonrpcc [janssonrpc_io.c:218]: fail_request(): Request timeout: #012{"id":3,"jsonrpc":"2.0","method":"core.version"} But [root@sipchel kamailio]# curl -s -H 'Content-Type: application/json' --data-binary '{"jsonrpc":"2.0", "method": "core.version"}' http://localhost:5060/jsonrpc { "jsonrpc": "2.0", "result": "kamailio 5.0.3 (x86_64/linux) " } How to get working rpc client in kamailio? Is it possible to define url for janssonrpcc module? -- Aydar A. Kamalov

1 0

rtpproxy sdp
by Isravel Raja Thangamani 18 Sep '17

18 Sep '17

Hi Thanks in advance if anyone can point me in the correct direction . I have kamailio running in front of some asterisk. SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I'm running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider no audio from external caller to the asterisk ext neither asterisk to external caller I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs I think my mistake in somewhere in the cfg below. Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints? This e-mail message and any documents attached to it are confidential and may contain information that is protected from disclosure by various federal and state laws, including the HIPAA privacy rule (45 C.F.R., Part 164). This information is intended to be used solely by the entity or individual to whom this message is addressed. If you are not the intended recipient, be advised that any use, dissemination, forwarding, printing, or copying of this message without the sender's written permission is strictly prohibited and may be unlawful. Accordingly, if you have received this message in error, please notify the sender immediately by return e-mail or call (insert phone number), and then delete this message.

3 2

ringing calls dialogs
by Aidar Kamalov 18 Sep '17

18 Sep '17

Hello, is it possible from route get info about all calls in ringing state? (for call pickup implementation) Is it better to do via dialogs or presence modules? -- Aydar A. Kamalov

3 4

Issue in ACC_CDR
by Logeshwaran G 18 Sep '17

18 Sep '17

Hi, I have enabled both cnxcc and ACC cdr flags. The call is dropped when the credit is got over (its handling by cnxcc correctly.) In this case the details are not storing in the ACC_cdr table. I am not sure what I am missing. in cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails Error: 2(19484) ERROR: acc [acc_cdr.c:373]: write_cdr(): message is empty! 2(19484) ERROR: acc [acc_cdr.c:656]: cdr_on_end_confirmed(): failed to write cdr! Please Guid me. Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 1

Siremis - version from git don't work
by Yu Boot 18 Sep '17

18 Sep '17

Downloaded new Siremis version via "git clone" and replaced /var/www/ content by new files. Now any menu entry gives the following error: https://pastebin.com/hV8ngRVx If if copy old Siremis files back, everything works fine.

2 8

Polycom Presence Routing for PUBLISH Messages
by John Janke 18 Sep '17

18 Sep '17

So I'm using a Polycom VVX 310 with Presence enabled (both on Phone and Kamailio) and the phone is sending SUBSCRIBE messages but not PUBLISH. I have tried various phone configurations without success. The closest thread to my situation I found online said this: "The thing is Polycom phones perform end-to-end subscription to presence event. It worked when I changed the routing configuration to simply relay the SUBSCRIBE request to the monitored device. (I'm using OpenSER as SIP proxy.)" I'm not sure if this is correct as it's from 2012, but in the case that it is I don't quite understand how to implement this. Any help on how to edit the config file? John Janke Progressive Innovation and Automation, Inc. PO BOX 309 Plainfield, IL 60544 Direct: 847.316.9028 Office: 800.688.9051 x 55328 Fax: 630.860.1017

2 1

Re: [SR-Users] Siremis graph time format is a bug!?
by Humberto Rodríguez 18 Sep '17

18 Sep '17

Thanks Daniel-Constantin Mierla, I followed your recommendation, and began to use a date column as INT. Then, I started to insert the date as unix_timestamp, like this (unix_timestamp(now())), which was the only way the graphic showed the date correctly (Active_Failed_Dialogs_4_hours - From 2017-09-15 12:04:01 To 2017-09-15 16:00:01), I failed inserting (now()+0), siremis didn't understand this format too. Cheers, Hto Now No Matter How Hard The Past, You Can Always Begin Again ~ -------------------------------------------- On Thu, 9/14/17, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Subject: Re: [SR-Users] Siremis graph time format is a bug!? To: "Humberto Rodríguez" <humberj77(a)yahoo.es>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, sr-users(a)lists.sip-router.org Date: Thursday, September 14, 2017, 4:09 AM Hello, try to set the type of the time column to int, so it is retrieved as numeric timestam, not as formated datetime. Cheers, Daniel On 13.09.17 05:39, Humberto Rodríguez wrote: Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had siremis 4.2 and never saw it correctly, I just updated to Siremis 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file <?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- <ChartGroup name="sumdialogs"> <Chart name="sumdialogs" table="active_dialogs" title="Active_Failed_Dialogs" type="area" order="reverse" orderby="ORDER BY id DESC" limit="LIMIT 10080"> <XAxis>  <Item name="date_created" data="date_created" title="date_created" type="timestamp"/> </XAxis> <YAxis> <Item name="active_dialogs" data="active_dialogs" title="active_dialogs" type="dataset" color="#D02020"/> <Item name="failed_dialogs" data="failed_dialogs" title="failed_dialogs" type="dataset" color="#DFC329"/> </YAxis> </Chart> </ChartGroup> --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" ------------------------------------------------------------------------------------------ # id, date_created, active_dialogs, early_dialogs, expired_dialogs, failed_dialogs, processed_dialogs, '1', '2017-04-11 14:40:40', '220', '0', '1219', '37', '6879491' '2', '2017-04-11 14:42:02', '206', '0', '1219', '37', '6879604' '3', '2017-04-11 14:43:01', '215', '0', '1219', '37', '6879685' '4', '2017-04-11 14:44:01', '191', '0', '1220', '37', '6879744' '5', '2017-04-11 14:45:02', '203', '0', '1220', '37', '6879837' '9', '2017-04-11 14:46:01', '211', '0', '1220', '37', '6879929' '11', '2017-04-11 14:47:01', '234', '0', '1220', '37', '6880019' '12', '2017-04-11 14:48:01', '222', '0', '1220', '37', '6880087' ---------------------------------------------------------------------------------------------------------------------------------- No Matter How Hard The Past, You Can Always Begin Again ~ _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 1

Kamailio Offline Messages
by Jango 17 Sep '17

17 Sep '17

I need help about offline messages this is my full working kamailio.cfg with audio/video call & instant messaging but i need features to receive user offline messages. kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_NAT #!define WITH_TLS #!define WITH_ACCDB # # Kamailio (OpenSER) SIP Server v4.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="jsrver.tk" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # -- for siremis CDRs -------------- loadmodule "rtimer.so" loadmodule "sqlops.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "//etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@localhost/kamailio") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # ====================================================== # Populate CDRs Table of Siremis # ====================================================== route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 1

Dispatcher Module with ds_is_from_list function
by Ilie Soltanici 17 Sep '17

17 Sep '17

Hello, "ds_is_from_list" function from Dispatcher module give false result - when actually it should be true. modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") request_route { xlog("L_INFO","[=Start Routing=]") if(ds_is_from_list()) { xlog("L_INFO","[=FROM ASTERISK=]"); } else { xlog("L_INFO","[=TO ASTERISK=]"); } if($si=="192.168.1.10") { xlog("L_INFO","[=FROM ASTERISK BY IP=]"); } else { xlog("L_INFO","[=TO ASTERISK BY IP=]"); } root@kamailio:/var/log# kamctl dispatcher show dispatcher gateways +----+-------+---------------------+-------+----------+-------+-----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+---------------------+-------+----------+-------+-----------------+ | 1 | 1 | sip:192.168.1.10:5060 | 0 | 3 | | Main Asterisk | | 2 | 2 | sip:192.168.1.11:5060 | 0 | 2 | | Backup Asterisk | | 3 | 3 | sip:192.168.1.12:5060 | 0 | 1 | | Test Asterisk | +----+-------+---------------------+-------+----------+-------+-----------------+ root@kamailio:/var/log# cat /var/log/kamailio.log [=Start Routing=] [=TO ASTERISK=] [=FROM ASTERISK BY IP=] Additional Information - SIP Packet 2017/09/15 19:21:57.589451 192.168.1.50:5061 -> 192.168.1.1:5060 REGISTER sip:192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5061;branch=z9hG4bK-5219-1-0 From: sip:10@192.168.1.50;tag=1 To: sip:10@192.168.1.50 Call-ID: 1-5219(a)192.168.1.50 CSeq: 1 REGISTER Contact: sip:10@192.168.1.50:5061 Max-Forwards: 10 Expires: 120 User-Agent: SIPp/Win32 Content-Length: 0 - Kamailio Version - output of kamailio -v root@kamailio:/var/log# kamailio -v version: kamailio 5.0.3 (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 12:12:14 Sep 7 2017 with gcc 6.3.0 root@kamailio:/var/log# - Operating System: Debian 9.1/ Linux kamailio1 4.9.0-3-amd64 #1 SMP Debian 4.9.30-2+deb9u3 (2017-08-06) x86_64 GNU/Linux Is this a bug or something wrong with configuration? Thank you

2 1

Implementing C or Java file in kamailio
by Logeshwaran G 15 Sep '17

15 Sep '17

Hi, Is there any way to implement my own C or Java program in kamailio.cfg to Validate my own scenarios? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 2

Replacing the From header user part.
by vivek＠advaitamtech.com 15 Sep '17

15 Sep '17

Hi All, Am trying to replace the user part of From header with own user name in kamailio.cfg I tried to assign directly, Eg: $fU = "abcd" Sometime when I receive From header as ' "12345" <12345(a)1.1.1.1>;tag=xxxx '. In this case it works as ' "12345" <abcd(a)1.1.1.1>;tag=xxxx ' . So i tried using replace_all() function. But this ended up adding extra characters(in my case "abcd") infront of content_type header. I dont know why. Can someone please help me out in replacing the from header in this case? Thanks, Vivek

2 2

Maximum Registration on Kamailo??
by amit Kumar 14 Sep '17

14 Sep '17

Hi All, kamctl ul show => show below result, does that mean max 1024 contact can be registered, if that is the case how to increase the max registration limit. Domain:: location table=*1024 *records=2 max_slot=1 Thanks

3 2

kamailio - multiple asterisk and chanspy
by Жан Базаров 14 Sep '17

14 Sep '17

Hello! how i can use asterisk's chanspy_app in kamailio-loadbalance-asterisk system?

2 1

Killing Calls for a Profile
by Sunil More 14 Sep '17

14 Sep '17

Hello All, Is there a method / command to kill calls related to one dialog profile. Right now i have to find all dialogs for a customer and loop through all the dialog hashes to run kamcmd dlg.end_dlg till i can kill all calls. Is there a easier way to do this. Thanking You, Sunil More Ph : 9503338275

2 4

Siremis "Chat Service" time format has bug or is only config issue??
by Humberto Rodríguez 14 Sep '17

14 Sep '17

Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had version 4.2 and never saw it correctly, I just updated to version 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file ?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- > <ChartGroup name="sumdialogs"> > > <Chart name="sumdialogs" table="active_dialogs" > title="Active_Failed_Dialogs" type="area" order="reverse" > > > orderby="ORDER BY id DESC" limit="LIMIT 10080"> > > <XAxis> >  > > <Item > name="date_created" data="date_created" title="date_created" > type="timestamp"/> > > </XAxis> > > <YAxis> > > <Item > name="active_dialogs" data="active_dialogs" > title="active_dialogs" type="dataset" color="#D02020"/> > > > <Item > name="failed_dialogs" data="failed_dialogs" > title="failed_dialogs" type="dataset" color="#DFC329"/> > > </YAxis> > </Chart> > </ChartGroup> > --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" > ------------------------------------------------------------------------------------------ > > > # id, date_created, active_dialogs, > early_dialogs, expired_dialogs, failed_dialogs, > processed_dialogs, > '1', '2017-04-11 14:40:40', '220', '0', > '1219', '37', '6879491' > '2', '2017-04-11 14:42:02', '206', '0', > '1219', '37', '6879604' > '3', '2017-04-11 14:43:01', '215', '0', > '1219', '37', '6879685' > '4', '2017-04-11 14:44:01', '191', '0', > '1220', '37', '6879744' > '5', '2017-04-11 14:45:02', '203', '0', > '1220', '37', '6879837' > '9', '2017-04-11 14:46:01', '211', '0', > '1220', '37', '6879929' > '11', '2017-04-11 14:47:01', '234', > '0', '1220', '37', '6880019' > '12', '2017-04-11 14:48:01', '222', > '0', '1220', '37', '6880087' > > ----------------------------------------------------------------------------------------------------------------------------------

2 1

Siremis graph time format is a bug!?
by Humberto Rodríguez 14 Sep '17

14 Sep '17

Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had siremis 4.2 and never saw it correctly, I just updated to Siremis 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file <?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- <ChartGroup name="sumdialogs"> <Chart name="sumdialogs" table="active_dialogs" title="Active_Failed_Dialogs" type="area" order="reverse" orderby="ORDER BY id DESC" limit="LIMIT 10080"> <XAxis>  <Item name="date_created" data="date_created" title="date_created" type="timestamp"/> </XAxis> <YAxis> <Item name="active_dialogs" data="active_dialogs" title="active_dialogs" type="dataset" color="#D02020"/> <Item name="failed_dialogs" data="failed_dialogs" title="failed_dialogs" type="dataset" color="#DFC329"/> </YAxis> </Chart> </ChartGroup> --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" ------------------------------------------------------------------------------------------ # id, date_created, active_dialogs, early_dialogs, expired_dialogs, failed_dialogs, processed_dialogs, '1', '2017-04-11 14:40:40', '220', '0', '1219', '37', '6879491' '2', '2017-04-11 14:42:02', '206', '0', '1219', '37', '6879604' '3', '2017-04-11 14:43:01', '215', '0', '1219', '37', '6879685' '4', '2017-04-11 14:44:01', '191', '0', '1220', '37', '6879744' '5', '2017-04-11 14:45:02', '203', '0', '1220', '37', '6879837' '9', '2017-04-11 14:46:01', '211', '0', '1220', '37', '6879929' '11', '2017-04-11 14:47:01', '234', '0', '1220', '37', '6880019' '12', '2017-04-11 14:48:01', '222', '0', '1220', '37', '6880087' ---------------------------------------------------------------------------------------------------------------------------------- No Matter How Hard The Past, You Can Always Begin Again ~

2 1

Cannot msg or call
by Костик 14 Sep '17

14 Sep '17

Hello, I have setup Kamailio 5.0.3 under Ubuntu 16.04. I can connect to the server but cannot call/text my contacts and don't see them online. When I try to call them, I get errors like ERROR: <core> [core/tcp_main.c:4264]: tcpconn_main_timeout(): connect 10.8.8.218:51079 failed (timeout) For texts, the error is 405 Method not allowed What is the problem and how to fix it? Thanks Serj

2 2

kamailio 4.4 registering to a SIP trunk provider using UAC then outbound routing to it.
by Jonathan Hunter 13 Sep '17

13 Sep '17

Hi Guys, I have a SIP provider who only accept registrations from users, so I have registered my kamailio Server with them using the uac module. It is registered fine, and I can get inbound calls on the trunk. My query is the best way to send outbound calls out via the registered sip gateway? I would normally load balance using dispatcher if it was to IP authenticating providers, but as they are using registration, what would be best practice to send outbound calls via the registration which is establised with the uac module? Is it just a question of doing it within the configuration file, as again they will look to auth requests as well with a 401/407 so I assume I can use uac_auth() for that. Just wondering what other people have done if anything, is my other option is to register using a B2Bua on the same server, and leave kamailio just as a proxy. Many thanks Jon

2 2

dialog keep-alive don't work
by Yu Boot 13 Sep '17

13 Sep '17

Hello. I'm using following dialog module settings: modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "profiles_with_value", "limithash ; caller") modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 300) modparam("dialog", "send_bye", 1) and I expect some OPTIONS within dialogs. But that don't happen. I don't see any OPTIONS messages in sngrep in established calls. What's wrong? I call dlg_manage() right before main t_relay, everything including CDRs are working fine...

2 3

Issue in Acc module
by Logeshwaran G 13 Sep '17

13 Sep '17

Hi, I am getting the below Error while Call Ended. 2(28504) ERROR: acc [acc_cdr.c:373]: write_cdr(): message is empty! 2(28504) ERROR: acc [acc_cdr.c:656]: cdr_on_end_confirmed(): failed to write cdr! Am I missing something? Please Guide me Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

1 1

kill the call using callid
by Logeshwaran G 13 Sep '17

13 Sep '17

Hi, Is it possible to kill the call using callid, I have referred the dialog module, I found the command: kamcmd dlg.end_dlg 342 56 But I need to kill the call with Callid, Input Please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 3

Can not register on kamailio
by Adesh Pandey 13 Sep '17

13 Sep '17

I am new to kamailio and I want to setup Kamailio as dispatcher for two Asterisk boxes but SIP registrations must be happening on the Kamailio itself. I have enabled mysql module and used auth module but when I try to register it says unauthorized access. In subscribers list I have added an entry for sip user 1 with domain as kamailio IP. Please help me to understand where can I improve so that I can implement register on Kamailio. -- Adesh Pandey Sr. Software Developer [image: phone] +91 92129 92129 Ext: 56 [image: moible] +91 8527384897 [image: Profile Pic] [image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter] <https://twitter.com/MyOperator>[image: Twitter] <https://www.youtube.com/user/MyOperatorCo>[image: Linkedin] <https://www.linkedin.com/company/voicetree-technologies>[image: Google+] <https://plus.google.com/+MyoperatorCo/posts> [image: Logo] <http://myoperator.co/>

2 4

Problem implementing usrloc via dmq
by Aleksandar Sosic 13 Sep '17

13 Sep '17

Hi everyone, we're trying to make work usrloc over dmq without a db on our nodes. Our `kamailio.cfg` code to make it work on the nodes is like this: ``` ... loadmodule "usrloc.so" ... loadmodule "dmq.so" loadmodule "dmq_usrloc.so" ... modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:routers.ourcloud.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) ``` Where `DMQ_ADDRESS="sip:<some_address>:5060"` is the public IP address of the running node. `nslookup` of `routers.ourcloud.com`: ```➜ ~ nslookup routers.ourcloud.com Server: 8.8.8.8 Address: 8.8.8.8#53 Non-authoritative answer: Name: rrouters.ourcloud.com Address: <IP1> Name: routers.ourcloud.com Address: <IP2> ``` and obviously IP1 and IP2 are pubblic IPs of kamailio nodes that respond to my pings. When i register a client to node IP1 it's ul dump is: ``` { "jsonrpc": "2.0", "result": { "Domain": "location", "Size": 1024, "AoRs": [{ "Info": { "AoR": "0409828030", "HashID": 1970914268, "Contacts": [{ "Contact": { "Address": "sip:0409828030@172.19.0.255:56328;ob", "Expires": 289, "Q": -1, "Call-ID": "5mclwiX4oYAVK0z-iT7M0p9RBX5-02g6", "CSeq": 59427, "User-Agent": "Telephone 1.2.6", "Received": "sip:<PROXY_IP>:5060", "Path": "[not set]", "State": "CS_NEW", "Flags": 0, "CFlags": 192, "Socket": "udp:<IP1>:5060", "Methods": 8159, "Ruid": "uloc-599d8fa8-f-1", "Instance": "[not set]", "Reg-Id": 0, "Last-Keepalive": 1503498176, "Last-Modified": 1503498176 } }] } }], "Stats": { "Records": 1, "Max-Slots": 1 } }, "id": 1 } ``` On node with IP2 the ul dump is: ``` { "jsonrpc": "2.0", "result": { "Domain": "location", "Size": 1024, "AoRs": [], "Stats": { "Records": 0, "Max-Slots": 0 } }, "id": 1 } ``` What are we doing wrong? Are we missing something? -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

3 7

Help to understand kamailio error msgs
by José Seabra 12 Sep '17

12 Sep '17

Hello there, I'm facing some error msg on my kamailio server in production that i would like to understand what they means in order to avoid them. The msg are: - *ERROR: tm [t_reply.c:1270]: t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200* - *dispatcher [dispatch.c:1509]: ds_load_replace(): cannot find load for (16720.mydomain.com <http://16720.mydomain.com>)* - *dispatcher [dispatch.c:2270]: ds_next_dst(): cannot update load distribution* - *tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 487 reply when a final 603 was sent out* - *tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 487 reply when a final 480 was sent out* - *WARNING: tm [t_lookup.c:245]: ack_matching(): WARNING: ack_matching() attempted on a transaction with no E2EACK callbacks => the results are not completely reliable when forking is involved* Thank you Best regards -- José Seabra

2 3

jsonrpc execution error
by David Villasmil 12 Sep '17

12 Sep '17

Hello guys, i configured jsonrpc on kamailio.cfg and RPCFIFOPATH on kamctlrc to the same file, seems to work for some commands, but not for others: root@ip-172-31-30-160:/home/admin# kamctl trusted show +----+--------------+-------+--------------+--------------+- -----+----------+ | id | src_ip | proto | from_pattern | ruri_pattern | tag | priority | +----+--------------+-------+--------------+--------------+- -----+----------+ | 1 | 1.2.3.4 | udp | | NULL | | 0 | | 2 | 5.4.3.2 | udp | | NULL | | 0 | +----+--------------+-------+--------------+--------------+- -----+----------+ root@ip-172-31-30-160:/home/admin# kamctl trusted dump { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32022 } root@ip-172-31-30-160:/home/admin# kamctl trusted reload { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32041 } Any help? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Multiple Htable Reload
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, I have multiple Htables, Is it possible to Reload all the Htables at one time via jsonrpc: As of Now I am using the below command to Reload one htable at a time. {"jsonrpc": "2.0","method": "htable.reload","params": ["b"],"id": 123} Input Please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 1

Map Technique(Key value pair) in kamailio
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, Is there any way to use the Mapping technique in kamailio. For Example: ========== Key = 91, Value = India Key = 92, Value = Srilanka How to store Key Value Pairs into kamailio and how to Retrieve? If I pass 91 it should Retrieve India, Is it possible? Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 8

Passive Open TCP sessions increases and kamailio runs out of memory
by Niklas Larsson 11 Sep '17

11 Sep '17

Hi, I have a kamailio 4.4.5 (x86_64/linux), running as a proxy in front of asterisk, handling udp/tcp/tls registration, using udp towards asterisk. Over time memory is consumed and the Passive Open TCP sessions is increasing until we run out of memory and a restart is needed. Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f9d55184000, 34120); Free fragment not found! Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f9d55184000, 34120) called from tls: tls_init.c: ser_malloc(275), module: tls; Free fragment not found! This image contains graphs that show the behaviour: https://imgur.com/68PvmyA Should it behave like this? Is there a knob to twist to change behaviour? tcp / tls stuff from kamailio.cfg: enable_tls=yes tcp_max_connections=8196 tls_max_connections=8196 tcp_no_connect=yes # Don't try to connect - drop if there is no session tcp_connection_lifetime=3605 modparam("tcpops", "closed_event", 1) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) in the registrar route I have: tcp_keepalive_enable("60", "5", "5"); /niklas

1 0

kamailio asterisk queue
by Жан Базаров 11 Sep '17

11 Sep '17

hello! how i can forward presence peer state to asterisk queue? Thank's

2 1

Q: purpose of t_newtran() in kamailio-basic.cfg route[LOCATION]?
by Anthony Alba 09 Sep '17

09 Sep '17

In kamailio-basic.cfg route[LOCATION] we have the following: --snip-- $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: --snip-- What is the purpose of t_newtran() here? Since we have not hit t_relay() yet (and don't have a transaction) is this to ensure the caller sees/acks the 404/405? Earlier in the mailing list https://lists.kamailio.org/pipermail/sr-users/2015-April/088062.html I read, although this is in reply to a different situation: "Why do you need to create new transactions yourself with t_newtran()? Nonexotic applications of Kamailio usually don't require this. ..." # User location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($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; } Thanks AAlba

2 1

Re: [SR-Users] append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

After I add route(DISPATCH) to branch_route branch calls are going throught asterisk # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); xnotice("BRANCH: $rm $ru [$ci]"); route(DISPATCH); route(NATMANAGE); } but if I set $ru = "sip:34200@domain.org"; append_branch("sip:34201@domain.org"); first main ($ru) INVITE resolve to contact of ip phone INVITE sip:34200@172.16.3.7:5060 but branch doesnt' resolve INVITE sip:34201@domain.org Is there any way to resolve it? -- Aydar A. Kamalov

2 1

append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

Hello! I have a simple route route[HUNT] { $ru = "sip:34200@172.16.3.7:5060"; append_branch("sip:34201@192.168.2.71"); } 172.16.3.7 and 192.168.2.71 ip phones 192.168.10.54 is kamailio 192.168.10.47 is asterisk If I called to this route kamailio will call to 34200 through dispatcher via asterisk. But branch going directly to the ip phone. Is it possible to call all branches through dispatcher and asterisks? U 192.168.10.54:5060 -> 192.168.10.47:5060 INVITE sip:34200@172.16.3.7:5060 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.0. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. U 192.168.10.54:5060 -> 192.168.2.71:5060 INVITE sip:34201@192.168.2.71 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.1. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. -- Aydar A. Kamalov

2 1

need some help about acc module
by Vidja Prashant 08 Sep '17

08 Sep '17

Hello, I have some feature required changes in acc module. In acc module we getting cdrs in database only which have METHOD=INVITE METHOD=BYE but in my setup I also get METHOD=OPTIONS so I need this entry also in acc table in kamailio database. so which changes i need to do in kamailio acc module.... or other...... my cdrs log I paste below which have mehod=options "Sep 7 07:16:53 sip /usr/local/sbin/kamailio[29264]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: request accounted: timestamp=1504783013;method=OPTIONS;from_tag=eUC9UgGrD;to_tag=j90H6yy5pU6Xc;call_id=2xJd0KCt-P;code=200;reason=OK;calling=46566464645;called=you;egressIP= kamailio.org;src_ip=1.0.0.127;to=sip:+918866213320@sip.call.org ;tag=j90H6yy5pU6Xc;from=sip:918888888888@localhost;contact=;pai=;uri=sip: you(a)kamailio.org;onnet=0;rate=0.000333333" Thanks.. # #

2 1

crash after using async_route()
by Vitaliy Aleksandrov 07 Sep '17

07 Sep '17

Hello kamailio list, Recently found a problem in my configuration that uses async_route() functionality. It crashes after several calls when wait_timer fires. #0 0xb74a8556 in raise () from /lib/libc.so.6 #1 0xb74a9d78 in abort () from /lib/libc.so.6 #2 0x08293ae2 in qm_free (qmp=0xad65d000, p=0x3d64692d, file=0xb6216a16 "tm: h_table.c", func=0xb621663c <__FUNCTION__.18751> "free_cell_helper", line=187, mname=0xb621664d "tm") at core/mem/q_malloc.c:471 #3 0xb613f103 in free_cell_helper (dead_cell=0xae2cd210, silent=0, fname=0xb6239ea5 "timer.c", fline=655) at h_table.c:187 #4 0xb61e7758 in wait_handler (ti=557858937, wait_tl=0xae2cd258, data=0xae2cd210) at timer.c:655 #5 0x0826a2cc in timer_list_expire (t=557858937, h=0xad6b9668, slow_l=0xad6ba144, slow_mark=312) at core/timer.c:874 #6 0x08267cb1 in timer_handler () at core/timer.c:939 #7 0x0826a4d3 in timer_main () at core/timer.c:978 #8 0x08069575 in main_loop () at main.c:1721 #9 0x080707ca in main (argc=11, argv=0xbf85f044) at main.c:2723 When crash happens, kamailio prints the following message: Sep 4 16:15:38 [18938]: : <core> [core/mem/q_malloc.c:469]: qm_free(): BUG: qm_free: bad pointer 0x70707553 (out of memory block!) called from tm: h_table.c: free_cell_helper(187) - aborting Also had a few crashes in retransmission_handler(): #0 0xb750b556 in raise () from /lib/libc.so.6 #1 0xb750cd78 in abort () from /lib/libc.so.6 #2 0xb6249b5a in retransmission_handler (r_buf=0xae036674) at timer.c:367 #3 0xb6247558 in retr_buf_handler (ticks=1234464444, tl=0xae036688, p=0x1f40) at timer.c:594 #4 0x0826a2cc in timer_list_expire (t=1234464444, h=0xad71c668, slow_l=0xad71cd44, slow_mark=2232) at core/timer.c:874 #5 0x08267cb1 in timer_handler () at core/timer.c:939 #6 0x0826a4d3 in timer_main () at core/timer.c:978 #7 0x08069575 in main_loop () at main.c:1721 #8 0x080707ca in main (argc=11, argv=0xbff64134) at main.c:2723 ERROR: tm [timer.c:366]: retransmission_handler(): transaction 0xae0365e0 scheduled for deletion and called from RETR timer (flags 6d) Both timers fired for an INVITE transaction that was previously suspended by async_route(), then resumed, sent out and received a 4xx reply (407). This configuration worked fine with kamailio 4.2.x and problem appeared after upgrading to 5.0.2. Trying to figure out how to narrow down the problem. Any input is appreciated.

2 6

alias_db muliple usernames for alias
by Aidar Kamalov 07 Sep '17

07 Sep '17

Hello, I have modparam("alias_db", "append_branches", 1) and table dbaliases +----+----------------+-----------------+----------+-----------------+ | id | alias_username | alias_domain | username | domain | +----+----------------+-----------------+----------+-----------------+ | 1 | 1111 | sipchel | 34200 | sipchel | | 2 | 1111 | sipchel | 34201 | sipchel | +----+----------------+-----------------+----------+-----------------+ If I call to 1111 ringing only 34200, because first invite is right (resolving to username contact), but branch inivite is wrong. Is it bug or it is my my fault? First: INVITE sip:34200@172.16.3.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.0 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> Second: INVITE sip:34201@sipchel SIP/2.0 Record-Route: <sip:192.168.10.54;lr=on;ftag=1898fba2a5729040o0;did=958.fd11> Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.1 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> -- Aydar A. Kamalov

1 0

handle_tcpconn_ev(): connect failed
by Donat Zenichev 07 Sep '17

07 Sep '17

Hi. Recently I've come across with TCP connection problem. The topology is as following: DNS srv load balancer - two kamailio proxy servers - one routing server. Client appeals to NAPTR record like: sip.domain.com So dns returns one of the proxy servers to client (depending on weight/priority). Now both kamailio have the same priority and weight (the goal is load balancing). Routing server (now it is asterisk) working with chan_pjsip.so, that supports NAPTR/SRV records. He is able to resolve Record-Route / Route headers with value - sip.domain.com (that proxy servers add to record-route headers while relaying requests to him). This topology is done to support present dialogs, even if proxy that recently processed it, is dead. But the problem comes, when routing server (asterisk) sends in-dialog requests to the proxy, that wasn't used to establish the dialog. Example, routing server obtains 200 OK from endpoint (relayed by kamailio1 to him) and he sends back ACK, but not to the kamailio1, he sends it to kamailio2 (because he resolves NAPTR sip.domain.com and gets ip of second kamailio). Kamailio2 processes the request as usual, because both kamailio have the same db for dialog module, but when he tries to relay the request to endpoint, he gots the error: ERROR: <core> [tcp_main.c:4070]: handle_tcpconn_ev(): connect XXX.XXX.XXX.XXX:52185 failed The port that kamailio2 tries to use to relay the ACK, is port that endpoint used to establish the dialog with kamailio1 and actually his TCP connection is now established with kamailio1. So kamailio2 tries to use the same port and gets the error. And this is proper behavior I think. There is no problem with UDP transport. Has anyone seen the similar problem? That indeed is not a problem, but proper behavior. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

2 4

cr failure routes
by Nelson Migliaro 07 Sep '17

07 Sep '17

Hello, I am using carrierroute in Kamailio (great module btw). All calls go the same gateway (rewrite_host). I am routing over differents vendor using a prefix. Prefix: 21 dnis: 999999999 $rU after rewrite_host is 21999999999 My problem is when trying to use a failure route, next domain use a different prefix but the host. So I get the: Candidate destination <gateway> was previously used Is there a flag or something to prevent this verification ? Thank you Nelson.-

2 1

How can client get registered users list from Kamailio server or server may broadcast this actively?
by Jack Wang 07 Sep '17

07 Sep '17

Hey guys, As indicated by the title, is there any module providing the related function? My server version is 4.2. Thanks. :)

2 1

Cost deduction in kamailio
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, How to deduct the call cost or credits for the subscribers with the cnxcc module? Input Please Thanks & Kind Regards, Logeshwaran G

3 15

Query in LCR
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, I am handling Two Different Account holders and Two Different Gateways with the different lcr_ids: id :1 lcr_id:1 gw_name:car1 ip_addr : 1.2.3.4 id :2 lcr_id:2 gw_name:car2 ip_addr : 1.2.5.3 In kamailio config file I am using load_gws(1), Here Comes my doubt, I need to Pass the lcr_id value inside the load_gws(lcr_id) Dynamically, Is it Possible , Please Guide me. Thanks & Kind Regards, *Logeshwaran G* *"Be Hunger, Stay Foolish"*

2 2

Doubt in Lcr_table
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, May I know the Difference between the "prefix" column in Lcr_gw table and lcr_rule table. I am using the prefix value in lcr_rule table, like 00091. But What is the use of Prefix Column in lcr_gw table. Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hunger, Stay Foolish"*

2 2

Loading keys with key_type="1" from database for htable - Module params help needed
by George Diamantopoulos 06 Sep '17

06 Sep '17

3 3

Call pickup with feature code
by Cyrille Demaret 06 Sep '17

06 Sep '17

Hi, I'm trying to implement call pickup with a feature code for phones that doesn't support "Replaces" header. User has to dial for example *8 to pickup a call who is in the same group pickup. My idea is to loop for each members of the group (coming from a SQL table using sqlops) and check for calls in early state. I was thinking to use the dialog module to search for active calls but I don't find any function allowing to do this. Does anyone know a module with a function allowing to find an active calls by callee? Thank you, Cyrille

1 0

SQL for ordered LCR rules
by Patrick Wakano 06 Sep '17

06 Sep '17

Hello list, Hope you all doing well! I am trying to ease the management of LCR routing rules, since once we begin to have multiple prefixes, multiple GWs and so on, the visualization and management of the rules priorities becomes exponentially hard to do. So first thing I am trying to achieve is an easy way of retrieving the rules in an ordered manner. I couldn't find any tool to do such thing and source code was not very friendly.... so I've come up with this Postgresql query that I think retrieves all rules in the same order I expect LCR to select the GWs. SELECT lr.lcr_id, lr.prefix, lrt.priority, lg.gw_name, lg.ip_addr FROM lcr_rule lr JOIN lcr_rule_target lrt ON lrt.lcr_id = lr.lcr_id AND lrt.rule_id = lr.id JOIN lcr_gw lg ON lg.lcr_id = lr.lcr_id AND lg.id = lrt.gw_id WHERE lr.enabled = 1 AND lg.defunct = 0 AND lr.lcr_id = ID AND lr.prefix SIMILAR TO '(|PREFIX%)' ORDER BY lr.lcr_id, LENGTH(lr.prefix) DESC, lrt.priority; It is missing the weights calculation, but it is rather complex and I am not using it anyway.... Other than that does anyone did something similar to check if my query really matches what LCR engine does? Thanks, Patrick Wakano

4 3

Re: [SR-Users] SQL for ordered LCR rules
by Patrick Wakano 06 Sep '17

06 Sep '17

Thanks for the response guys! The link https://github.com/dimitri/prefixIt is returning 404.... Regarding the performance itself I am not worried since this select it is just for management and I don't expect having millions of rules. The idea is just to have an easy way to have a picture of how the LCR will order and select the gateways based on a given prefix. The three LCR tables are not so easy to handle and manage from command line so my idea was to have a single SELECT or VIEW to return me all I need at once! >From what I could check, I think the select I sent pretty much translates what LCR module does internally, I am just trying to verify if it has some flaw, which could mislead me in the rules management. Cheers, Patrick Wakano On 6 September 2017 at 00:32, Dmitry Sinina <dmitry.sinina(a)onat.edu.ua> wrote: > https://yeti-switch.org/demo.html > > > On 9/5/17 5:29 PM, Dmitry Sinina wrote: > >> And you can try our opensource LCR engine. We use kamailio as load >> balancer and SEMS as SBC. >> >> On 9/5/17 3:02 AM, Patrick Wakano wrote: >> >>> Hello list, >>> >>> Hope you all doing well! >>> I am trying to ease the management of LCR routing rules, since once we >>> begin to have multiple prefixes, multiple GWs and so on, the visualization >>> and management of the rules priorities becomes exponentially hard to do. >>> So first thing I am trying to achieve is an easy way of retrieving the >>> rules in an ordered manner. I couldn't find any tool to do such thing and >>> source code was not very friendly.... so I've come up with this Postgresql >>> query that I think retrieves all rules in the same order I expect LCR to >>> select the GWs. >>> >>> SELECT lr.lcr_id, lr.prefix, lrt.priority, lg.gw_name, lg.ip_addr >>> FROM lcr_rule lr >>> JOIN lcr_rule_target lrt ON lrt.lcr_id = lr.lcr_id AND lrt.rule_id = >>> lr.id <http://lr.id> >>> JOIN lcr_gw lg ON lg.lcr_id = lr.lcr_id AND lg.id <http://lg.id> = >>> lrt.gw_id >>> WHERE lr.enabled = 1 AND lg.defunct = 0 AND lr.lcr_id = ID AND lr.prefix >>> SIMILAR TO '(|PREFIX%)' >>> ORDER BY lr.lcr_id, LENGTH(lr.prefix) DESC, lrt.priority; >>> >>> It is missing the weights calculation, but it is rather complex and I am >>> not using it anyway.... Other than that does anyone did something similar >>> to check if my query really matches what LCR engine does? >>> >>> Thanks, >>> Patrick Wakano >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >

2 4

real-time update question
by Stent, Bob 05 Sep '17

05 Sep '17

I am just starting out with Kamailio. One of the things I want to be able to do is add a domain to the configuration and have Kamailio pick it up without having to restart. I see there is a runtime update framework but I am having trouble navigating the documentation to find out what this actually does for me. Can somebody point me to the right set of docs to read to figure this out? Do I simply add an alias to the kamalio.cfg file and somehow send an RPC command to Kamalio to re-read configuration? Bob Stent Aspect Software 5 Technology Park Drive Suite #9 Westford, MA 01886 This email (including any attachments) is proprietary to Aspect Software, Inc. and may contain information that is confidential. If you have received this message in error, please do not read, copy or forward this message. Please notify the sender immediately, delete it from your system and destroy any copies. You may not further disclose or distribute this email or its attachments.

3 2

5.0 Version compiling gives warning
by Yasin CANER 05 Sep '17

05 Sep '17

3 3

Dialplan Module
by Nelson Migliaro 05 Sep '17

05 Sep '17

Hello, Do you know if its possible to

2 2

Re: [SR-Users] Incorrect value when setting number like values in Htable using Kamcmd
by Tony - Telecomy 05 Sep '17

05 Sep '17

Prefixing the numbers with "s:" fixed the issue. Thank you for your help. Tony

1 0

Incorrect value when setting number like values in Htable using Kamcmd
by Tony - Telecomy 05 Sep '17

05 Sep '17

Hello, I'm having an issue setting phone numbers values in htable using kamcmd when the phone number is above the integer range 2147483647. Initially I planned to set the values as strings but this is not working as expected as well. Please find below some examples: root@tst:~/tst/test# kamcmd htable.seti tst_sim integer_limit "2147483647" root@tst:~/tst/test# kamcmd htable.seti tst_sim integer_exceeded_limit "2147483648" root@tst:~/tst/test# kamcmd htable.sets tst_sim integer_as_string_valid "2147483647" root@tst:~/tst/test# kamcmd htable.sets tst_sim integer_as_string_invalid "2147483648" root@tst:~/tst/test# kamcmd htable.dump tst_sim | grep -A 1 integer name: integer_as_string_valid value: 2147483647 -- name: integer_limit value: 2147483647 -- name: integer_as_string_invalid value: 18446744071562067968 -- name: integer_exceeded_limit value: -2147483648 my ctl params are as follow: modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") modparam("ctl", "binrpc_max_body_size", 1024) modparam("ctl", "binrpc_struct_max_body_size", 1024) Thanks, Tony

3 2

telcosuite3-crm sql
by Kordován Szabolcs 05 Sep '17

05 Sep '17

Hi, I would like to try the IMS according to the Kamailio World 2017: Kamailio For IMS And VoLTE presentation. Unfortunately I have problem with telcosuite, the sql structure is missing. Where can I find the whole create.sql? Thanks for help. Regards, Szabolcs

3 7

Kamailio SIP protocol over WSS
by Washington Costa 05 Sep '17

05 Sep '17

Hi, I solved my problem. I was missing load some modules and routers. I saw this file as example: https://gist.github.com/jesusprubio/4066845 but is important to know that every module the begin "mi" they aren't supporte on Kamailio 5.0. You will need replace for the module relative the version 5.0. Washington L da Costa Analista de Sistemas (82) 9 9823-4782

2 1

Question about fix_nated_contact.
by Володимир Іванець 04 Sep '17

04 Sep '17

Hello everyone, I have system setup where there are few Asterisk servers behind Kamailio that is behind the firewall. There is a problem with call initiation. When phone from outside sends INVITE to Kamailio, it then passes it to one of Asterisk servers. Asterisk server eventually replies to Kamailio with 200 OK. Contact header contains Asterisk's local IP address and port. Kamailio executes function *fix_nated_contact* before sending reply to the phone. However Contact header will still have Asterisk's local socket. *fix_nated_contact* is being executed with no doubt. I can see xlog output that I set before and after it. Can someone tell how exactly *fix_nated_contact* gets "*request's source address:port pair*" and if it can be printed to the log? Thank you!

1 0

Homer confusion ....
by Mark Boyce 04 Sep '17

04 Sep '17

Hi All Trying to get a basic version of homer working on Kamailio V5. So far I have sipcapture throwing everything in to a single mysql table. So first question; Should I be using sipcapture or siptrace? I’m not sure I can see the difference. Second question; On what I have working so far using sipcapture, I’m seeing the invites, subscribes, cancels, act’s all end up in the database. What I’m missing is any responses from Kamailio back to the handset (AUTh Challenges etc) modparam("sipcapture", "db_url", "mysql://sipcapture:secretpassword@localhost/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "insert_retries", 5) modparam("sipcapture", "insert_retry_timeout", 10) plus a sprinkling of "sip_capture(); “ in the “xxx_route {“ blocks What am I missing? Mark

4 6

presence_xml mod_init(): Can't bind xcap_client
by Borislav Gergov 04 Sep '17

04 Sep '17

*Hi I have presence_xml parameter force_active set to 1 and still i get **error** presence_xml:mod_init: Can't bind xcap_client* *kamailio 5.1.0-dev6 (x86_64/linux) *

2 3

MWI forwarding between Kamailio and Freeswitch
by Wang, Wallace 04 Sep '17

04 Sep '17

Hello We're using Kamailio & freeswitch as my company voice system. So far, everything is good, except the MWI. I tried to find out how to forward MWI between Kamailio and Freeswitch, or let the Presence_MWI module can work with the freeswitch behind Kamailio. If it was possiable, could anyone give me an idea or some links or demo configure for this. I really appreciate your help. Best Regards Wallace Wang

2 1

uac +kamailio + dispatcher + asterisk
by Жан Базаров 04 Sep '17

04 Sep '17

Hello! provider trunks registration on kamailio UAC?! it's clear. but how does asterisk find out through which trunk the call should be made? I tried to set the header from asterisk dialplan. for example: if (is_method("INVITE")) { #record_route_preset("109.195.102.122"); route(DIRECTION); setflag(FLT_ACC); # do accounting } # ------ LOADBALANCE ROUTE ------------ # if(!ds_is_from_list()) { route(DISPATCH); } route[DISPATCH] { #round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(NATMANAGE); route(DRELAY); exit; } route[DIRECTION] { if ($hdr(x-trunk) != $null) { if (!is_method("BYE")){ $fu=""; t_on_failure("MANAGE_FAILURE"); $dlg_ctx(timeout_route) = "DIALOG_END"; $avp(i:10)=43200; $dlg_ctx(timeout_bye) = 0; sql_pvquery("ca", "select l_uuid, auth_username, auth_password, realm, l_domain, r_domain from uacreg where id='$hdr(x-trunk)'", "$avp(uuid), $avp(uname), $avp(passwd), $avp(realm), $avp(src_ipaddr), $avp(dst_ipaddr)"); t_on_failure("MANAGE_FAILURE"); $dlg_ctx(timeout_route) = "DIALOG_END"; $avp(i:10)=43200; $dlg_ctx(timeout_bye) = 0; $fu=""; uac_replace_from("sip:$avp(uname)@$avp(dst_ipaddr)"); $tu="sip:"+$tU+"@"+$avp(dst_ipaddr); $ru="sip:"+$tU+"@"+$avp(dst_ipaddr); remove_hf("Contact"); $var(contact)="sip:"+$avp(uname)+"@10.49.9.2:5060"; insert_hf("Contact: <$var(contact)>\r\n"); #insert_hf("Contact: "); msg_apply_changes(); fix_nated_register(); xlog("L_INFO","Contact header $var(contact) 111111111111111111111111111111111111111 is $ct {$ct}\n"); route(RELAY); } #### BYE } ### XTRUNK But in this configuration I do not come bye but when I register providers trunks on asterisk - problem with BYE not visible. but I can not register provider-trunks on all the asterisks, because incoming invite arrives at the link + address, and all the asteriscs ring. All my asterisk's behind nat

2 1

jsonrpcs execution error
by David Villasmil 04 Sep '17

04 Sep '17

Hello guys, i configured jsonrpc on kamailio.cfg and RPCFIFOPATH on kamctlrc to the same file, seems to work for some commands, but not for others: root@ip-172-31-30-160:/home/admin# kamctl trusted show +----+--------------+-------+--------------+--------------+------+----------+ | id | src_ip | proto | from_pattern | ruri_pattern | tag | priority | +----+--------------+-------+--------------+--------------+------+----------+ | 1 | 1.2.3.4 | udp | | NULL | | 0 | | 2 | 5.4.3.2 | udp | | NULL | | 0 | +----+--------------+-------+--------------+--------------+------+----------+ root@ip-172-31-30-160:/home/admin# kamctl trusted dump { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32022 } root@ip-172-31-30-160:/home/admin# kamctl trusted reload { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32041 } Any help? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Carrierroute, uac_*, failure_route.
by yu＠yu-boot.ru 04 Sep '17

04 Sep '17

Hello. Have a following task. Select carrier by carrierroute module from db, try to route (dial, relay, bridge - you name it) call there and if call is failed/rejected due to any reason, go to failure_route(?), perform carrierroute lookup again for another carrier and route call there. Main problem is that different carriers require different From/To format. And Kamailio don't allow to use uac* functions in failure_route. How to resolve this? Maybe entire routing method is wrong for my task? Any advice? -- sent from myMail for Android

3 3

Re: [SR-Users] [Freeswitch-users] Fwd: Call drops after 32 seconds with ACK Timeout
by Karsten Horsmann 02 Sep '17

02 Sep '17

Hi Mykyta, since you are plain sip without encryption you can see your sip-traffic on the wire, use sngrep on the kamailio box to figure out whats with the sip-dialogs - it helps a lot. https://github.com/irontec/sngrep and if its more an kamailio config problem, it would be better to stay at sr-users@. Or you can setup an webhomer for debugging. Kind Regards 2017-09-01 17:49 GMT+02:00 Sean Ingham <sean(a)missionlabs.co.uk>: > Once a call is answered FreeSWITCH sends a 200 OK SIP message to your > Kamailio loadbalancer. As per the SIP spec your kamailio box should respond > to a 200 OK with an ACK. The FS logs show that it never receives an ACK, > and it repeatedly tries sending the 200 OK with no response. After 30 > seconds without an ACK FS will assume the call has failed and hangup. > > Looking at your Kamailio SIP logs you can see the 200 OKs getting through > okay, and no attempt to send an ACK back to FS. I think this indicates an > issue with your kamailio.cfg rather than FreeSWITCH. > > > Sean > www.wirex-precision.co.uk > > On Fri, Sep 1, 2017 at 3:40 PM, Микита Нікітенко < > nikitanikitenko(a)gmail.com> wrote: > >> >> Hello! I have server at digital ocean with a floating ip address and I >> have routed sip traffic to this floating ip. At this server installed >> kamailio which balancing between freeswitch servers. I have a problem that >> call drops after 32 seconds with error ACK Timeout. Traffic comming throug >> the kamailio proxy. >> >> kamailio.cfg - https://pastebin.com/x0CL3TEZ >> kamailio sip messages log - https://pastebin.com/wyT3cBbD >> freeswitch logs with sip debug - https://pastebin.com/zPysfQkV >> >> I'm struggling for a week and have no idea how to fix it. >> Thank you very much. >> Mykyta Nikitenko >> >> >> >> -- Mit freundlichen Grüßen *Karsten Horsmann*

2 1

Call drops after 32 seconds with ACK Timeout
by Микита Нікітенко 01 Sep '17

01 Sep '17

Hello! I have server at digital ocean with a floating ip address and I have routed sip traffic to this floating ip. At this server installed kamailio which balancing between freeswitch servers. I have a problem that call drops after 32 seconds with error ACK Timeout. kamailio.cfg - https://pastebin.com/x0CL3TEZ kamailio sip messages log - https://pastebin.com/wyT3cBbD freeswitch logs with sip debug - https://pastebin.com/zPysfQkV I'm struggling for a week and have no idea how to fix it. Thank you very much. Mykyta Nikitenko

2 1

Change on the fly logging verbosity - Kamailio v5.0.2
by Karsten Horsmann 01 Sep '17

01 Sep '17

Hello, at this moment i try to figure out the best way to add an "switch" for debug/log output. I use many xlog, like everyone that starts with kamailio too. Then i go further and add #!ifdef cases around to switch this via #!define. Is there any other way to turn on/off my "debug"-logging on the fly, while kamailio is still running? #!define TESTBED_MODE request_route{ ... #!ifdef TESTBED_MODE route(DEBUG); #!endif ... #!ifdef TESTBED_MODE xlog("L_INFO","---FIX_SIPS-after: ru:($ru)"); #!endif ... } -- Kind Regards *Karsten Horsmann*

3 3

Kamailio v5.0.3 Released
by Daniel-Constantin Mierla 01 Sep '17

01 Sep '17

Hello, Kamailio SIP Server v5.0.3 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.0.x versions are strongly recommended to be upgraded to v5.0.3. For more details about version 5.0.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/09/kamailio-v5-0-3-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.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

last pkg.stats entry
by Juha Heinanen 01 Sep '17

01 Sep '17

When I make pkg.stats query (latest master), I get a weird last entry. What is it? -- Juha ... { entry: 28 pid: 14675 rank: -4 used: 9450336 free: 2354944 real_used: 10227968 total_size: 12582912 total_frags: 310 } { entry: 29 pid: 0 rank: -128 used: 0 free: 0 real_used: 0 total_size: 12582912 total_frags: 0 }

2 5

Planning Kamailio v5.0.3
by Daniel-Constantin Mierla 01 Sep '17

01 Sep '17

Hello, I am considering to release Kamailio v5.0.3 sometime next week, therefore it is the time to backport patches with fixes from master that apply to 5.0 branch. If you are aware of such fix that has to be backported, reply here or make a pull request on github project. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 3

Kamailio SIP protocol over WSS
by Washington Costa 31 Aug '17

31 Aug '17

Hello, I'm using kamailio 5.0.2 (x86_64/linux). And it works with WS. Regards, Washington Costa

1 0

Kamailio SIP protocol over WSS
by Washington Costa 31 Aug '17

31 Aug '17

Hi. I'm trying use the JSSIP cliente and Kamailio with WSS. On these sites I generated my certificates: https://letsencrypt.org/ and https://certbot.eff.org/ My browser identify the certificates but SIP client just say: failed: WebSocket opening handshake was canceled Do you have any idea how to fix it? Do you can say a checklist with everything I need to do for use Kamailio with WSS? Here is my log on Kamailio: Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 187.41.60.10 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:985]: tcpconn_new(): on port 44630, type 3 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:1291]: tcpconn_add(): hashes: 3062:1194:1491, 2 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa4ac60, 53, 2, 0x7f259df9cd80), fd_no=44 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa4ac60, 53, -1, 0x0) fd_no=45 called Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:4139]: handle_tcpconn_ev(): sending to child, events 1 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3818]: send2child(): selected tcp worker 1 26(13225) for activity on [tls:172.31.39.220:4443], 0x7f259df9cd80 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1574]: handle_io(): received n=8 con=0x7f259df9cd80, fd=9 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: tls [tls_server.c:226]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7f259ddc4a80 ctx 0x7f259de03bc0 sn []) Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL handshake started Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_main.c:2430]: tcpconn_do_send(): sending... Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7f259df9cd80 n=2568 fd=9 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_main.c:2465]: tcpconn_do_send(): buf= Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: #026#003#001 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa93f00, 9, 2, 0x7f259df9cd80), fd_no=1 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:195]: tcp_emit_closed_event(): TCP closed event creation triggered Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:202]: tcp_emit_closed_event(): no callback registering for handling TCP closed event - dropping! Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:305]: tcp_read_data(): EOF on 0x7f259df9cd80, FD 9 ([187.41.60.10]:44630 ->26(13225) DEBUG: <core> [core/tcp_read.c:306]: tcp_read_data(): -> [172.31.39.220]:4443) Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1341]: tcp_read_req(): EOF Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa93f00, 9, -1, 0x10) fd_no=2 called Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1495]: release_tcpconn(): releasing con 0x7f259df9cd80, state -1, fd=9, id=2 ([187.41.60.10]:44630 -> [187.41.60.10]:4443) Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1499]: release_tcpconn(): extra_data 0x7f259dfb6488 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3250]: handle_tcp_child(): reader response= 7f259df9cd80, -1 from 1 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: tls [tls_server.c:665]: tls_h_close(): Closing SSL connection 0x7f259dfb6488 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 187.41.60.10 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:985]: tcpconn_new(): on port 44692, type 3 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:1291]: tcpconn_add(): hashes: 2869:1129:1296, 3 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa4ac60, 53, 2, 0x7f259df9cd80), fd_no=44 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa4ac60, 53, -1, 0x0) fd_no=45 called Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:4139]: handle_tcpconn_ev(): sending to child, events 1 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3818]: send2child(): selected tcp worker 2 27(13226) for activity on [tls:172.31.39.220:4443], 0x7f259df9cd80 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1574]: handle_io(): received n=8 con=0x7f259df9cd80, fd=9 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: tls [tls_server.c:226]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7f259ddc4a80 ctx 0x7f259de05708 sn []) Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL handshake started Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_main.c:2430]: tcpconn_do_send(): sending... Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7f259df9cd80 n=2568 fd=9 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_main.c:2465]: tcpconn_do_send(): buf= Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: #026#003#001 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa93f00, 9, 2, 0x7f259df9cd80), fd_no=1 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:195]: tcp_emit_closed_event(): TCP closed event creation triggered Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:202]: tcp_emit_closed_event(): no callback registering for handling TCP closed event - dropping! Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:305]: tcp_read_data(): EOF on 0x7f259df9cd80, FD 9 ([187.41.60.10]:44692 ->27(13226) DEBUG: <core> [core/tcp_read.c:306]: tcp_read_data(): -> [172.31.39.220]:4443) Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1341]: tcp_read_req(): EOF Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa93f00, 9, -1, 0x10) fd_no=2 called Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1495]: release_tcpconn(): releasing con 0x7f259df9cd80, state -1, fd=9, id=3 ([187.41.60.10]:44692 -> [187.41.60.10]:4443) Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1499]: release_tcpconn(): extra_data 0x7f259dfb6488 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3250]: handle_tcp_child(): reader response= 7f259df9cd80, -1 from 2 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: tls [tls_server.c:665]: tls_h_close(): Closing SSL connection 0x7f259dfb6488

2 1

question for best way to remove sips uris - TLS and sips in $ru - get_send_socket2(): protocol/port mismatch
by Karsten Horsmann 31 Aug '17

31 Aug '17

Hello List, of course - best way is not to use sips: uri scheme. But i have to deal with that. I try to configure an multihomed kamailio (public/private IP) in front of my SIP-Servers. One of my softphones variants used TLS for connection but the INVITES of this softphone use the "sips" URI scheme. As i saw in other posts, this seems to be an issue for many people. In my case it breaks the routing to my internal sip-server (only plain udp). Most of my config is similar to havfos example + TLS. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg I tried to fixed that with this extra route - but this breaks the dialog somewhere - after 60 seconds i have no audio and the BYE becomes another CSeq: in the request_route right after route(AUTH) i add my route(FIX_SIPS) if (is_method("INVITE|SUBSCRIBE")) { if ($ru =~ "^sips:") { route(FIX_SIPS); } else { record_route(); } } route[FIX_SIPS] { if ($ru =~ "^sips:") { xlog("L_INFO","---FIX_SIPS-before: ru:($ru)"); $var(orig_uri) = $ru; $ru = "sip:" + $rU + "@" + $td; xlog("L_INFO","---FIX_SIPS-now: $rU $td ru:($ru) orig_uri:($var(orig_uri))"); record_route(); #$ru = $var(orig_uri); xlog("L_INFO","---FIX_SIPS-after: ru:($ru)"); } } 172.20.120.81 = Kamailio private IP 172.20.120.79 = My internal SIP-Server (B2BUA) I used kamailio with CentOS 7 and RTPENGINE 63bdd615 from master kamailio-5.0.2-3.1.x86_64 kamailio-tls-5.0.2-3.1.x86_64 kamailio-mysql-5.0.2-3.1.x86_64 kamailio-websocket-5.0.2-3.1.x86_64 -- Kind Regards *Karsten Horsmann*

1 2

sql_do_query_async(): the db driver module doesn't support async query
by Juha Heinanen 31 Aug '17

31 Aug '17

sqlops README tells: 4.5. sql_query_async(connection, query) Make an async SQL query using 'connection', if implemented by db driver module (e.g., db_mysql) my sqlops sqlcon parameter specifies mysql driver, but still i get to syslog: Aug 24 08:02:05 rox1 /usr/bin/sip-proxy[13886]: ERROR: sqlops [sql_api.c:427]: sql_do_query_async(): the db driver module doesn'tsupport async query any idea why? -- juha

2 7

kamailio asterisk + provider sip trunks
by Жан Базаров 31 Aug '17

31 Aug '17

Good morning! kamailio + dispatcher + provider trunks. How to make RR for incoming calls from pstn? my trunks are authorized on an asterisk, but i have problem with multiple calls on each asterisk. If I do through uac then I have a problem with BYE, and in general the module is uncomfortable. Please tell me how to implement a bunch: kamailio + asterisk + provider trunks..

2 1

Kamailio failover - dialog (transaction) that is available on different proxies
by Donat Zenichev 30 Aug '17

30 Aug '17

Hi. I came up with idea to set up stand with two kamailio and one b2bua server (for routing). The idea consists of failover for dialogs, transactions. So if one of kamailio nodes is down, another one is able to catch up the dialog and let users to properly end up the session. For better realizing of it, I will try to describe the idea step by step: 1. UAC invites UAS, they've done three-way-handshake, media stream is up. 2. Kamailio that processed this dialog is down. 3. Users decided to end the session with BYE method, but proxy that processed their three-way-handshake recently is down, so one of ua sends BYE to the destination route that contains domain name (that both kamailio serve), BYE achieves the second kamailio to let him properly end the dialog. But, there is a big but, this second kamailio hasn't ever known about this dialog, he doesn't support any transactions for it and furthermore he doesn't know anything about this call-id. So the solution for it, as I think, is hidden in db mode for user location (columns that contain call-ids, branches etc. But I need to be sure, if I'm on the right way. For purpose, where one ip is served by two nodes, I have two solutions: -First one. I want to create heartbeat cluster with two kamailio nodes, they will have one shared ip address, so when one node gets down, another one brings up shared ip interface and implements the same actions that master does. -Another method is to assign a few ip addresses to one domain name (ip addresses of different kamailio proxies). So the goal looks simple, if someone has ever done something like that, I will be glad to read the ideas. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

3 5

Simultaneous calls in kamailio
by Logeshwaran G 30 Aug '17

30 Aug '17

Hi, Is there any way(tools) to place Simultaneous Calls in kamailio: For Example: 1000 calls simultaneously in Kamailio. Input Please?? Thanks & Kind Regards, Logeshwaran G

4 6

Fwd: sdp extra \r\n in multipart body
by Diego Nadares 29 Aug '17

29 Aug '17

Hi Daniel! Thanks for your reply. The rtpengine module I think it's expecting two \r\n like: ...pcma/8000\r\n \r\n --uniqueboundary .... cpc,9\r\n \r\n --uniqueboundary-- I saw the code and I modified it but Can any one cofirm if multipart has to be like ...pcma/8000\r\n \r\n --uniqueboundary Or ...pcma/8000\r\n --uniqueboundary Or any one of those? Thanks again! Diego El El lun, 21 de ago. de 2017 a las 02:33, Daniel-Constantin Mierla < miconda(a)gmail.com> escribió: > Hello, > > there is \r\n after > > a=rtpmap:8 PCMA/8000 > > because > > --uniqueBoundary-- > > is on next line. > > Maybe is the fault of the code extracting sdp from multi-part body. Open > an issue on github bug tracker not to forget about it. If nobody looks at > it, I will try to do it, but it may take few days given traveling. > > Cheers, > Daniel > > On 18.08.17 22:21, Diego Nadares wrote: > > Hi guys, > > I have the following multipart body > > Content-Type: multipart/mixed;boundary=uniqueBoundary > Remote-Party-ID: <tel:1112222;phone-context=+54> > <1112222;phone-context=+54> > Content-Length: 364 > > --uniqueBoundary > Content-Type: application/sdp > > v=0 > o=user1 53655765 2353687637 IN IP4 111.111.111.111 > s=- > c=IN IP4 111.111.111.111 > t=0 0 > m=audio 6000 RTP/AVP 8 > *a=rtpmap:8 PCMA/8000* > *--uniqueBoundary--* > Content-Type: application/gtd > Content-Disposition: signal;handling=optional > > IAM, > CPC,20 > GCI,885sdfasdf1123 > > --uniqueBoundary-- > > > As you can see *--uniqueBoundary *before *a=rtpmap:8 PCMA/8000* has not > an extra '\r\n'. Is this correct? Or should have it? > > The thing is that rtpengine is receiving sdp with no final '\r'n' from > kamailio: > > Dump for 'offer' from 127.0.0.1:37388: { "sdp": "v=0#015#012o=user1 > 53655765 2353687637 IN IP4 111.111.111.111#015#012s=-#015#012c=IN IP4 > 111.111.111.111#015#012t=0 0#015#012m=audio 6000 RTP/AVP 8#015#012*a=rtpmap:8 > PCMA/8000"*, "ICE": "remove", "direction": [ "pub", "priv" ], "replace": > [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", > "call-id": "1-17755(a)111.111.111.111", "received-from": [ "IP4", > "111.111.111.111" ], "from-tag": "1", "command": "offer" } > > And the resulting sdp from rtpengine is this: > > --uniqueBoundary > Content-Type: application/sdp > > v=0 > o=user1 53655765 2353687637 IN IP4 172.16.213.16 > s=- > c=IN IP4 172.16.213.16 > t=0 0 > m=audio 35066 RTP/AVP 8 > *a=rtpmap:8 PCMA/8000a=sendrecv* > a=rtcp:35067 > > The problem is related with that '\r\n' but Is this a malformed message or > a possible bug? > > Thanks in advance! > > Diego > > > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > >

2 3

Query in LCR Rule Target Table:
by Logeshwaran G 29 Aug '17

29 Aug '17

Hi, I need to know What is the Purpose of Priority and Weight column in Lcr_rule_target table: Example: | id | lcr_id | rule_id | gw_id | priority | weight | status | createdBy | createdDate | modifiedBy | modifiedDate | +----+--------+---------+-------+----------+--------+--------+-----------+-------------+------------+--------------+ | 1 | 1 | 1 | 1 | 1 | 1 | 1 | NULL | NULL | NULL | NULL | | 2 | 1 | 4 | 1 | 1 | 2 | 1 | NULL | NULL | NULL | NULL Kindly someone please Guide me. Thanks & Kind Regards, Logeshwaran G

2 4

LCR dont_strip_or_prefix_flag
by Patrick Wakano 29 Aug '17

29 Aug '17

Hello list, I am using Kamailio 5.0.2 on centos7 and I am trying to use the dont_strip_or_prefix_flag ( https://www.kamailio.org/docs/modules/5.0.x/modules/lcr.html#idp49131644) for LCR routing but I discovered that just by declaring the flag like this the module won't prefix my calls: #!define LCR_DONT_STRIP_OR_TAG 7 modparam("lcr", "dont_strip_or_prefix_flag", LCR_DONT_STRIP_OR_TAG) It doesn't matter if I set or reset the flag it never prefix the ReqUR if the declaration is present. I have to remove the declaration to make the module prefix it correctly. But then the whole purpose of the flag is lost....Anyone aware of this behaviour? Regards, Patrick Wakano

3 4

TSILO + NAT issues
by Emanuele Gambaro 28 Aug '17

28 Aug '17

Hi to all. I have the following setup-needs : *) Kamailio 5.0.1 with TSILO module and Apple push notification service. *) Rtpengine configured in order to force a "media-handover” for every call. *) NAT forced for all connection. I have a lot of problem with the INVRESUME route. When an INVITE is resumed (upon mobile client connection via APN message) the NAT does not work, and there is no rtp flow between devices. It works only if device is already online (so without the SUSPEND route). How can I setup TSILO in order to force NAT management for all calls? Thank you -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

2 1

lcr gws
by Logeshwaran G 28 Aug '17

28 Aug '17

Hi, I am having two lcr gateways in lcr_gw table. But When executing the "kamcmd lcr.dump_gws", Its showing only one gateway list: [root@zeodialer ~]# kamcmd lcr.dump_gws { lcr_id: 1 gw_id: 1 gw_index: 1 gw_name: Carrier1 scheme: sip: ip_addr: 104.251.178.29 hostname: port: 5060 params: transport: ;transport=udp strip: 0 prefix: tag: flags: 1 state: 0 defunct_until: 0 } Did I missing something? Thanks & Kind Regards, Logeshwaran G

2 8

Fwd: Weird issue with kamailio relaying messages to itself
by George Diamantopoulos 26 Aug '17

26 Aug '17

Stupid gmail replying to sender only... ---------- Forwarded message ---------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: 27 August 2017 at 02:29 Subject: Re: [SR-Users] Weird issue with kamailio relaying messages to itself To: Daniel-Constantin Mierla <miconda(a)gmail.com> Hello all, I've figured out what was going on, so I'm sharing here in case anyone else runs into this. I guess it should be a fairly common situation under certain circumstances when using the dispatcher module... The problem was that no destinations in the dispatcher set used were available for these requests. So $du was not set by ds_select_next. Which meant that when t_relay() was called later in the script, it would route based on R-URI, and the RURI's uri was kamailio itself. As to the reason why I was left with no dispatcher destinations available, well, I would mark destinations offline for 500 "server error" responses coming from them, and asterisk (which is the receiving application for all of dispatcher's destination sets) will send out a 500 to the B-leg when it receives a 480 to an A-leg. Getting a single 480 to one of the asterisk boxes would cause this to happen across all destinations, as kamailio would retry the next destination after a 500 failure and would receive a 500 from all of them in the end (because all of them would get the 480 in such small time frame). BR, George On 31 July 2017 at 21:52, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello Daniel, > > Thanks for the reply. That is correct, looping was not my intention, I'm > trying to figure out what's causing it... > > BR, > George > > On 31 July 2017 at 17:29, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> usually you should not loop requests locally, unless a very special case >> -- do you do the loop routing because of a need or just happens but you >> don't know why? >> >> Cheers, >> Daniel >> >> On 31.07.17 13:46, George Diamantopoulos wrote: >> >> Hello all, >> >> I have been toying with kamailio lately, and I thought I had gotten to >> the point where I had a mostly working (tm) prototype. I gave it a test >> drive with some real calls, however, and an issue manifested at least once, >> where homer received packets originating from the kamailio host, and whose >> destination was also the kamailio host. >> >> The dialog this manifested in is a generally problematic one, with many >> retransmissions occurring because of slow database access (I haven't >> implemented htable caching yet). No packet capture over the network is >> actually taking place, I'm copying everything to homer with "trace_mode"set >> to 1. >> >> Homer shows these messages like in the screenshot: >> https://imagebin.ca/v/3VGJAmovRBmo. Here's an example of a packet: >> >> >> 2017-07-28 14:32:29 +0300 : 2.3.4.5:5060 -> 2.3.4.5:5060 >> INVITE sip:1234567890@kamailio-server.org SIP/2.0 >> Record-Route: <sip:2.3.4.5;lr;ftag=as491cec82> >> Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bK0cf.0779 >> 9e3eb71f33a9ef91178ac760ebd2.0 >> Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bKsr-BnCVy >> rWoUladI14pU7Pry-Pzy-ONy-VSQ74NU7HzeYazq2nFU2Oc5FIWiGIKq-HXe >> x1oONpam-C6IrIXkr4FQxZRh7sM >> Max-Forwards: 69 >> From: <sip:subscriber@5.4.3.2:5061>;tag=as491cec82 >> To: <sip:1234567890@kamailio-server.org> >> Contact: <sip:2.3.4.5;line=sr-eNC05xhKefQnON1EglFrOkF0OfpzV2s9UzM9UXM >> NQXMn5-B0Q-s*> >> Call-ID: 4e5d22c3369704b97d866c8d0f3798f8@192.168.201.2:5061 >> CSeq: 103 INVITE >> User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1 >> Date: Fri, 28 Jul 2017 11:32:24 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Remote-Party-ID: "9876543210" <sip:9876543210@192.168.201.2> >> ;party=calling;privacy=off;screen=yes >> Content-Type: application/sdp >> Content-Length: 500 >> >> v=0 >> o=root 242468242 242468243 IN IP4 172.17.130.13 >> s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1 >> c=IN IP4 172.17.130.13 >> t=0 0 >> m=audio 59426 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:50 >> a=sendrecv >> a=rtcp:59427 >> a=ice-ufrag:hffIoy6x >> a=ice-pwd:nXB8ip7Qz8ZG5yyZRr97kGJbej >> a=candidate:igct4zWHEzhGCjWc 1 UDP 2130706431 <21%203070%206431> >> 172.17.130.13 59426 typ host >> a=candidate:igct4zWHEzhGCjWc 2 UDP 2130706430 <21%203070%206430> >> 172.17.130.13 59427 typ host >> >> Can something like this be triggered by misconfiguration in the routing >> scripts? Should it worry me and should I dig into it more? Could it be a >> bug of the siptrace module and nothing bad actually took place? I'm not >> sure where to start with this, so any input would be greatly appreciated. >> Thanks! >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - www.asipto.com >> Kamailio World Conference - www.kamailioworld.com >> >> >

2 1

Kamailio fails to process ACK to 487 from Cisco-SIPGateway/IOS-12.x?
by George Diamantopoulos 26 Aug '17

26 Aug '17

Hello all, I'm having a weird issue with Kamailio failing to properly process an ACK received to a 487 it sent, resulting in retransmissions of the 487. I assume it's because it can't match the ACK to the transaction, but I could be wrong. I'm using a modified version of the default configuration, so ACKs should be handled correctly. I haven't editted the WITHINDLG route in any way that would affect this (or at least I think). In addition, ACKs to 487 from other UAs are processed correctly, and these transactions are handled by the same routes in kamailio configuration as the problematic one, so I'm inclined to believe it's UA-specific? Here's an example transaction of the failed kind (results in kamailio retransmitting the 487): myself:5060 -> peer:5060 ------------------------- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP peer:5060 From: <sip:user@peer>;tag=116B5368-24D8 To: <sip:tel@myself>;tag=as655f6372 Call-ID: 84DC69F2-873811E7-8A639B5A-3D9194E8@peer CSeq: 101 INVITE Server: modCOM v2 SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 peer:49590 -> myself:5060 ------------------------- ACK sip:tel@myself:5060 SIP/2.0 Via: SIP/2.0/UDP peer:5060 From: <sip:user@peer>;tag=116B5368-24D8 To: <sip:tel@myself>;tag=as655f6372 Date: Wed, 23 Aug 2017 12:50:47 GMT Call-ID: 84DC69F2-873811E7-8A639B5A-3D9194E8@peer Max-Forwards: 10 Content-Length: 0 CSeq: 101 ACK And here's another similar transaction which is successful (no retransmissions): myself:5060 -> peer:5060 ------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP peer:5060;branch=z9hG4bKjbmvq4009gthskk0a6s1.1 From: <sip:user@anonymous.invalid;user=phone>;tag=599D7495-9ACE9E3-0A324A05 To: <sip:tel@anonymous.invalid:5060;user=phone>;tag=as65375e5d Call-ID: 599D7495-007A5832@fath3pcu238 CSeq: 1 INVITE Server: modCOM v2 SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 peer:5060 -> myself:5060 ------------------------ ACK sip:tel@myself:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP peer:5060;branch=z9hG4bKjbmvq4009gthskk0a6s1.1 From: <sip:user@anonymous.invalid;user=phone>;tag=599D7495-9ACE9E3-0A324A05 To: <sip:tel@anonymous.invalid:5060;user=phone>;tag=as65375e5d Call-ID: 599D7495-007A5832@fath3pcu238 Max-Forwards: 69 Content-Length: 0 CSeq: 1 ACK I can't pinpoint anything wrong with the first exchange, other than the fact that for some reason, the "less than" (<) sign in the from and to URIs is escaped as &lt in homer's GUI (which also breaks CSS rendering in Firefox, I had to clear this code out). However, these escaping characters are not visible with sngrep when capturing traffic normally, and neither when doing a select in homer's database directly, so I guess it's a rendering bug in homer-ui and can be ignored (unless someone has reason to believe otherwise). Now the relevant portion of the debug log is: Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) exec: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=223 a=24 n=t_check_trans Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:1001]: t_check_msg(): msg id=104 global id=103 T start=(nil) Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:459]: t_lookup_request(): start searching: hash=54992, isACK=1 Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:494]: t_lookup_request(): proceeding to pre-RFC3261 transaction matching Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:641]: t_lookup_request(): no transaction found Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:1070]: t_check_msg(): msg id=104 global id=104 T end=(nil) Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) exec: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=231 a=2 n=exit It explicitly states that no transaction is found, after initiating pre-RFC3261 (why?) transaction matching. However, even pre-3261 matching should work, as the from and to headers as well as call-id in request and repy are the same. Any input would be greatly appreciated, thanks! George

2 3

One Kamailio and many RTPENGINEs
by Denys Pozniak 25 Aug '17

25 Aug '17

Hello! I want to build distributed topology with one KAMAILIO and 5x RTPENGINEs for each region. I will have dedicated links between these sites, so I can garantee somehow voice quality. If one customer from region1 calls customer region2, RTP should go: Customer1->RTEPNGINE1->RTPENGINE2->Customer2 So the main question: Can one Kamailio build "call" between two RTPENGINEs ? BR, Denys

4 4

Handeling Proxy Cancel event
by Abdul Basit 25 Aug '17

25 Aug '17

Hi, I am trying to capture and distinguish two types of cancels generated by kamailio. 1- Cancel when Invite timeout 408 2- Cancel generated when call lands on multiple devices and any one of them answer the call. This type of cancel comes with (rfc3326) Reason: SIP;cause=200;text="Call completed elsewhere" How can I capture and isolate these proxy generated CANCEL events? -- regards, abdul basit

3 5

Diaplan module
by Nelson Migliaro 24 Aug '17

24 Aug '17

Hello, I am trying to use the dialplan module but I am having this problem: First of all, I pull the "dpid" value from the two first digits of the dnis: $var(dialplan_vendor) = $(rU{s.substr,0,2}); Then if(!dp_translate("$var(dialplan_vendor)","$dlg_var(servicio)/$var(ani_dialplan)")) { xlog("L_NOTICE","Error al seleccionar el dialplan\n"); sl_send_reply("403", "Not allowed"); drop; } But when I test everything I get: ERROR: dialplan [dialplan.c:233]: dp_get_ivalue(): no AVP, XAVP or SCRIPTVAR found (error in scripts) I think the problem is related to variable $var(dialplan_vendor). It is set up as string due to the transformation and dialplan need a integer. If I set up the variable this way for testing: $var(dialplan_vendor) = 18, it works correctly. Question, how can I transform the sting into a integer? Thank you Nelson.

1 1

Consistency: -1 sometimes true, sometimes false
by Sebastian Damm 24 Aug '17

24 Aug '17

Hi, we recently stumbled upon the following problem: We had something like this in our route: $var(ret) = allow_address("2"); if ($var(ret)) { do_something; } Turns out, this always evaluates to true, no matter whether the IP address really is in address group 2 or not. The values in $var(ret) are 1 or -1, depending on the outcome of allow_address(). However, if we write it this way if (allow_address("2")) { do_something; } the code inside is only executed if the IP address is in address group 2. Is this some kind of inconsistency? Should negative values in variables evaluate to true or false? Best Regards, Sebastian

3 5

Feature request: USRLOC jsonrpc ul.add - set all entry fields
by Dmitry Sytchev 23 Aug '17

23 Aug '17

Hi Daniel! Looking into code it seems RPC functions of USRLOC don't allow setting all fields for USRLOC entry, for example, Received and Socket. It would be great if we can set these, because we need to have ability to transfer contacts from other server to Kamailio. I think at present this can be done sending specially crafted fake REGISTER packets or calling special route from HTTP with some tricks for saving contact and received, but as for me it would be great to set it from RPC because we already have ul.add method. It also would be cool if we can use named parameters in JSONRPC params: {} section with names corresponding to usrloc fields, like 'table', 'AOR', 'received' and so on. Maybe on specual key not to mix with current 'params'. Thanks for your great work! -- Best regards, Dmitry Sytchev, IT Engineer

2 1

Feature request: USRLOC jsonrpc ul.add - set all entry fields
by Dmitry Sytchev 23 Aug '17

23 Aug '17

Hi Daniel! Looking into code it seems RPC functions of USRLOC don't allow setting all fields for USRLOC entry, for example, Received and Socket. It would be great if we can set these, because we need to have ability to transfer contacts from other server to Kamailio. I think at present this can be done sending specially crafted fake REGISTER packets or calling special route from HTTP with some tricks for saving contact and received, but as for me it would be great to set it from RPC because we already have ul.add method. It also would be cool if we can use named parameters in JSONRPC params: {} section with names corresponding to usrloc fields, like 'table', 'AOR', 'received' and so on. Maybe on specual key not to mix with current 'params'. Thanks for your great work! -- Best regards, Dmitry Sytchev, IT Engineer

1 0

Kamailio APP_LUA
by santosh gupta 23 Aug '17

23 Aug '17

Hi Robert, Hope you are doing good. You have helped me in the past. I need to set LCR via app_lua, getting rates and destination from mysql. Link which i am following is https://www.kamailio.org/wiki/embeddedapi/4.0.x/lua The issue is I am not able to get connect via sr commands for example sr.sqlops.query("connection", "query", "result") What should be exactly is "connection", "query", "result" Error I am getting in syslog is /usr/local/kamailio/etc/kamailio/kamailio.lua:1: attempt to index field 'sqlops' (a nil value) I am using kamailio 5.0.2 A little bit of headsup will help me. Thanks a lot again. -- Regards Santosh Gupta +91 9323043570 <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

2 2

Query in Dialplan module
by Logeshwaran G 23 Aug '17

23 Aug '17

Hi, I have one query in Dialplan module, my scenario is If I call 22291+++++++++ the first three digit should remove and the call place to 91+++++++. But i dont get the exact result, the first three digits are not removing. I have used the below rules: Lcr_rule table: ============ id: 5 lcr_id: 2 Prefix:22291 Request uri: NULL Stopper: 1 Enabled: 1 LCR_rule_target: ================== id:3 lcr_id:2 rule_id:5 gw:id:2 priority:1 weight:1 Dialplan table: ============= id:3 dpid:2 pr:1 match_op:1 match_exp: ^222[0-9]+$ subst_exp: ^222[0-9]+$ repl_exp:\1 Is it correct or not, Please guide me. Thanks & Kind Regards, Logeshwaran G

2 3

Set up caller ID
by Nelson Migliaro 23 Aug '17

23 Aug '17

Hello, I have several Asterisk sending calls to a centralizad Kamailio. I would like this Kamailio to set up the caller id number of the calls. I am using the uac_replace_from function but I would like to pull the caller id from a database to set up the number based on the sip vendor the call is forwarded What would be the correct way to do that? I was thinking of: sql_query("ca", "select ddi from ddi_table where xxx= $rU limit 1", "ra"); $avp(dnis) = $dbr(ra=>[0,0]); sql_result_free("ra"); But I am afraid of causing a bottheneck on the database. As usual, thank you for the help. Regards, Nelson.-

2 1

rtpproxy rtp timeout
by yu＠yu-boot.ru 23 Aug '17

23 Aug '17

Hello. What version of rtpproxy app should be used for this feature to work with Kami? How to use it correctly? -- Отправлено из myMail для Android

3 4

snmpstats get calls that have ended in 5xx cause code
by Olli Attila 23 Aug '17

23 Aug '17

Hello, I have Kamailio 5.0 configured with snmpstats module. I can succesfully query for eg. the current num of dialogs on kamailio by issuing: snmpget -v 2c -c public localhost KAMAILIO-MIB::kamailioCurNumDialogs.0 KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 Is it possible to make query about calls that have ended with 5xx sip cause code on last 5min or 15min? I noticed that the following is available on the snmpstats module but it only holds a counter that keeps increasing until I restart kamailio process: snmpget -v 2c -c public localhost KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 Also I think the above includes all cause codes that are between 3xx-6xx. Cheers, --Olli

1 0

cr reload in xmlrpc
by Logeshwaran G 23 Aug '17

23 Aug '17

Hi, It is the correct way of using the cr reload command in xmlrpc through postman? <?xml version='1.0'?> <methodCall> *<methodName>cr.reload</methodName>* </methodCall> Thanks & Kind Regards, Logeshwaran G

2 6

Save original $fU, $rU
by yu＠yu-boot.ru 22 Aug '17

22 Aug '17

Have a problem with Asterisk. When callee behind Asterisk hangs up first, it sends back BYE with swapped From and To fields, which affects Kami CDRs rendering them useless. I've tried to save original From/To to AVPs on initial INVITE but they are empty when CDR is written. How to solve this? -- Отправлено из myMail для Android

3 2

dp_translate_cr_route, R-URI
by Yu Boot 22 Aug '17

22 Aug '17

Hello. #cut off prefix "9" dp_translate("1"); ... #searching in db by called number without prefix... if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ ... and failing. Is it normal behavior that cr_route "sees" original $rU even after dp_translate manipulations?

1 0

Kamailio / Asterisk and multidomain
by Cyrille Demaret 22 Aug '17

22 Aug '17

Hi, I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration" tutorials, I would like to let Kamailio handle registrations, calls between users and other basic functionalities. Asterisk will only handle advanced features like voicemail, advanced dialplan configuration,. My problem is that It needs to be multidomain. My network will be as follow with 2 network interfaces for Kamailio and a private lan between Kamailio and Asterisk : Public interface ip 1.2.3.4 ----> [Kamailio] <----- Private interface 192.168.100.10 <----------------> 192.168.100.11 [Asterisk] When I send a call to asterisk, the domain is sent in the from field of the INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP domain using the ${SIPDOMAIN} variable). My problem is when I need to send a call back to Kamailio for example to reach another user of the domain. I'm using Asterisk 14 with PJSIP with the following config : [kamailio] type=endpoint transport=transport-udp context=from-kamailio disallow=all allow=ulaw aors=kamailio [kamailio] type=aor contact=sip:192.168.100.10:5060 [kamailio] type=identify endpoint=kamailio match=192.168.100.10 If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio /sip:200@testdomain.com", Asterisk contact directly testcomain.com without going through the local IP of my Kamailio. I can't send a domain to Kamailio in the INVITE request. Does anyone can help me on this or maybe simply tell me that I'm not going to the good direction? :) Thank you, Cyrille

3 3

Siremis - can't add carrier without prefix/suffix
by Yu Boot 22 Aug '17

22 Aug '17

Hello Suddenly revealed that I can't add carriers without rewrite_prefix/rewrite_suffix via Siremis. But I don't need these fields! And kamctl adds these rules fine. What's a problem? Screenshot with error http://tinypic.com/r/68v493/9

2 4

problems with use_dst_blacklist and ICMP "Destination Unreachable"
by Valentin Christoph 22 Aug '17

22 Aug '17

Hi all, Question: * Does the TM destination blacklist work, if destination host responds with ICMP "Destination Unreachable"? Discussion: I have got an IMS scenario, where I think the destination blacklist is not working and try to find the reason. I use UDP between two Kamailio SIP Proxies (P-CSCF and S-CSCF) and send a SIP request to the primary instance of the S-CSCF. However the destination Kamailio has been stopped with kamctl stop before and so the destination machine responds with ICMP "Destination Unreachable". DNS Failover towards secondary S-CSCF instance works perfectly. Destination Blacklist does not work. I.e. the next SIP Request is sent to the primary host again, before failover to secondary host applies. I would have expected the request would be sent to secondary host immediately. >From the description in doc/dst_blacklist.txt [...] A destination is added to the blacklist when an attempt to send to it fails (e.g. timeout while trying to send or connect on TCP), or when a SIP timeout occurs while trying to forward statefully an INVITE (using tm) and the remote side doesn't send back any response.[...] I think the destination blacklist does not work, because an ICMP Response "Destination Unreachable" is received. Second question: * How could I avoid the problem? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

rtpproxy: One direction rtp not being relayed
by Jon Bonilla (Manwe) 22 Aug '17

22 Aug '17

Hi all After a migration, a customer is reporting one way audio on incoming calls. A tcpdump capture shows 65 rtp packets in both directions but rtpproxy log shows RTP stats: 1 in from callee, 64 in from caller, 65 relayed, 0 dropped Any thoughts? cheers, Jon

2 1

sercmd dispatcher.list command giving an error: 500 - No Destination Sets
by Deepak Jaiswal 22 Aug '17

22 Aug '17

Hi, I am new to Kamailio and trying to debug an issue. We have two servers. On one server, I am getting destination like below. sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:xxx.xxx.xxx.139:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397149 } DEST: { URI: sip:xxx.xxx.xxx.:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397150 } } } However, on another server, I am getting below error. sercmd dispatcher.list error: 500 - No Destination Sets sercmd -s unixs:/tmp/kamailio_ctl dispatcher.list error: 500 - No Destination Sets [root@uk-vap-002 ~]# sercmd system.listMethods cfg.add_group_inst cfg.commit cfg.del cfg.del_delayed cfg.del_group_inst cfg.diff cfg.get cfg.help cfg.list cfg.rollback cfg.set cfg.set_delayed cfg.set_delayed_int cfg.set_delayed_string cfg.set_now_int cfg.set_now_string cfg.seti cfg.sets core.arg core.echo core.flags core.info core.kill core.printi core.prints core.ps core.psx core.pwd core.sctp_info core.sctp_options core.shmmem core.tcp_info core.tcp_options core.udp4_raw_info core.uptime core.version ctl.connections ctl.listen ctl.who dispatcher.list List going on.................................... I tried to find dispatcher.list file on both server. However, it is not there on both the server. Could you please help me on this? Thanks and Regards Deepak K. Jaiswal Tech support Engineer My Phone : +91- 95351-23352, +91- 88675-01385 My Movius work line: +1 (470) 298-7210 [ovius]<http://www.moviuscorp.com/> www.moviuscorp.com<http://www.moviuscorp.com/>

1 0

append_branch and tls
by James Cloos 21 Aug '17

21 Aug '17

[I see I posted this to -dev instead of -users; fixing that -JimC] When using rewrite_ruri() and append_branch() to set a list of next hops to try, what is the proper syntax to specify that a given route should use tcp or tls/tcp? I've mostly used a single next hop via t_relay_to_tcp() or t_relay_to_tls(), but need route stacks now. -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

2 2

Kamailio+SEMs conference call setting
by leslie mekiyess 21 Aug '17

21 Aug '17

Hi, I have a Kamailio server , a SEM one, an SQL server and Siremis installed and working. I am able to perform p2p calls with a softmobile. But I didn't succeed to perform conference / group / speed dial calls. Besides defining some records in the database, I don't know how to complete the necessary (Kamailio/SEM ...) configuration in order to perform conference call, group and speed dial . Any assitance would very appreciated. Leslie :-)

1 0

siptrace: how to duplicate the complete dialog?
by Daniel Tryba 21 Aug '17

21 Aug '17

I'm trying to trace a complete dialog on a loadbalancer/proxy: UAC <-> Proxy <-> backend 109.235.34.a <-> 109.235.32.b <-> 185.61.68.c Proxy runs 4.3.x (but with 4.4.x I appear to have the same results). siptrace config: ============================================= modparam("siptrace", "duplicate_uri", "sip:10.11.12.13:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) ============================================= After REQINIT (which does pike and handles OPTIONS), I setflag(22) the message. ============================================= route(REQINIT); setflag(22); route(NATDETECT); route(WITHINDLG); ============================================= Next depending in where the message comes from: ============================================= if(!is_in_subnet($si, backends)) { ds_select_dst(1, 0); t_on_failure("RTF_DISPATCH"); } else { remove_hf("Route"); loose_route(); } route(RELAY); ============================================= sip_trace() is being called in the routes: WITHINDLG MANAGE_BRANCH MANAGE_REPLY MANAGE_FAILURE RTF_DISPATCH The resulting duplicate packets are: 11:20:21.037 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.040 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.047 401 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:21.073 ACK 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.117 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.135 100 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:21.450 180 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:27.929 200 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:28.106 ACK 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:30.625 BYE 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:30.659 BYE 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:30.663 BYE 109.235.32.b:5060 109.235.34.a:39621 tcp 11:20:30.664 200 109.235.34.a 39621 109.235.32.b:5060 tcp 11:20:30.664 200 109.235.32.b:5060 185.61.68.c:5060 udp What is missing are the forwarded replies, there is an ACK that matches the 401, but I have no clue what message was send to the UAC. Adding a sip_trace after REQINIT shows nothing extra. How (if possible) do I get all messages being send via the proxy? I could use sngrep/sipgrep, but that is an other process to monitor for failure.

2 4

allow inbound calls from any ip
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, I need to allow the inbound calls from any ip i kamailio. Input please Thanks & Kind Regards, Logeshwaran G

2 1

Take call counts from kamailio
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, How to take call counts from the kamailio? Thanks & Kind Regards, Logeshwaran G

2 12

sdp extra \r\n in multipart body
by Diego Nadares 21 Aug '17

21 Aug '17

Hi guys, I have the following multipart body Content-Type: multipart/mixed;boundary=uniqueBoundary Remote-Party-ID: <tel:1112222;phone-context=+54> Content-Length: 364 --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 111.111.111.111 s=- c=IN IP4 111.111.111.111 t=0 0 m=audio 6000 RTP/AVP 8 *a=rtpmap:8 PCMA/8000* *--uniqueBoundary--* Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, CPC,20 GCI,885sdfasdf1123 --uniqueBoundary-- As you can see *--uniqueBoundary *before *a=rtpmap:8 PCMA/8000* has not an extra '\r\n'. Is this correct? Or should have it? The thing is that rtpengine is receiving sdp with no final '\r'n' from kamailio: Dump for 'offer' from 127.0.0.1:37388: { "sdp": "v=0#015#012o=user1 53655765 2353687637 IN IP4 111.111.111.111#015#012s=-#015#012c=IN IP4 111.111.111.111#015#012t=0 0#015#012m=audio 6000 RTP/AVP 8#015#012*a=rtpmap:8 PCMA/8000"*, "ICE": "remove", "direction": [ "pub", "priv" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "1-17755(a)111.111.111.111", "received-from": [ "IP4", "111.111.111.111" ], "from-tag": "1", "command": "offer" } And the resulting sdp from rtpengine is this: --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 172.16.213.16 s=- c=IN IP4 172.16.213.16 t=0 0 m=audio 35066 RTP/AVP 8 *a=rtpmap:8 PCMA/8000a=sendrecv* a=rtcp:35067 The problem is related with that '\r\n' but Is this a malformed message or a possible bug? Thanks in advance! Diego

2 1

Documenation of functions exported for KEMI
by Sebastian Damm 19 Aug '17

19 Aug '17

Hi, I was wondering whether there is some kind of documentation of all functions currently exported for KEMI by Kamailio modules. I know I can find them somehow via grep, but I could not find a page in the Kamailio wiki listing all of them. Is there any? Best Regards, Sebastian

2 1

Converting to kemi
by James Cloos 19 Aug '17

19 Aug '17

[I see I posted this to -dev instead of -users; fixing that -JimC] I'm working on converting some configs to kemi lua and have a couple of questions: First: Does KSR.pv.get() support multiple pseudos in a single call? The examples such as misc/examples/kemi/kamailio-basic-kemi-lua.lua make several calls, but also do things like KSR.pv.get("$T_branch_idx]") to avoid having the append "]" to the string. That makes me wonder why that file has code like this: KSR.dbg("new branch [".. KSR.pv.get("$T_branch_idx]") .. " to ".. KSR.pv.get("$ru") .. "\n"); instead of like: KSR.pv.get("[$T_branch_idx] to $ru\n") Second: Is it still the case that, eg, event_route[xhttp:request] cannot be in the kemi lua code? There was another. Perhaps I'll recall what it was... :( -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

2 1

Kamailio_428 memory usage
by Evgeniy G. 18 Aug '17

18 Aug '17

Good afternoon! Can you please tell us how the Kamailio works with shared memory. The fact is that there was an accident in which Kamailio reported an error - little common memory. This happened a day later as I added the dialplan module. After that, I began to monitor the values: used_size real_used_size max_used_size free And I saw that I can not properly assess. I note that the server does not process calls, it is idle. After the server is started, the indicators are as follows: used_size: 63 real_used_size: 146 max_used_size: 146 free 106 After rebooting the module (for example, drouting, in which the data has not changed): used_size: 63 real_used_size: 193 max_used_size: 231 free 62 I found, a description, the reasons for the increase in memory after rebooting the modules, I quote "As suggested by Daniel, for carrierroute reload, we need the same amount of memory as it used to be. Block when reloading the table. " But why: 1. after increasing the memory the server frees only real_used_size 2. That that is specified as real_used_size on my observations can not be released in current of several days (probably weeks). 3.If in this state again to reboot the module (the same one that led to an increase in memory), the memory does not increase. Question: during what time should kamailio release the memory specified as real_used_size? Also what is considered as real_used_size? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-428-memory-usage-tp161327.… Sent from the Users mailing list archive at Nabble.com.

1 0

syslog CDR time format
by Yu Boot 18 Aug '17

18 Aug '17

Hello. Can't get 'date-time' format timestamps in syslog CDRs, while SQL CDRs are OK. According to acc module documentation: modparam("acc", "cdr_enable", 1) modparam("acc", "time_format", "%Y-%m-%d %H:%M:%S") modparam("acc", "time_mode", 3) modparam("acc", "cdr_facility", "LOG_DAEMON") syslog: NOTICE: acc [acc_cdr.c:352]: log_write_cdr(): start_time=1503068984.627; end_time=1503068991.250; duration=6.623; src_user=100; src_domain=a.b.c.d; src_ip=e.f.g.h; dst_ouser=200; dst_user=200; dst_domain=k.l.m.n

1 0

On the fly Change
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any possibilities of "on the fly" option in kamailio.cfg. For Example: I need to change my carrier IP address in the lcr_gw table, kamailio.cfg should takes it on the fly. Without restarting the kamailio. Thanks & Kind Regards, Logeshwaran G

4 18

Acc cdr duration
by yu＠yu-boot.ru 18 Aug '17

18 Aug '17

Hello. How to remove milliseconds from duration field in CDR? -- Отправлено из myMail для Android

1 0

dispatch_rpc
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any other module need to be added for xmlrpc for using dispatch_rpc(). Because I am getting an below error: 0(22485) ERROR: <core> [core/cfg.y:3274]: yyparse(): cfg. parser: failed to find command dispatch_rpc (params 0) 0(22485) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_final.cfg, line 640, column 17: unknown command, missing loadmodule? I have loaded xmlrpc as well as xhttp modules. Do I need to enable some other modules as well?? Please Guide me. Thanks & Kind Regards, Logeshwaran G

1 0

SIP_DOMAIN in /etc/kamailio/kamctlrc
by Trent 17 Aug '17

17 Aug '17

Hello. I have not been able to find anything in the wiki, or documentation which covers this. The only item I have found is related to a module. >From the information, I have gleaned on this, it is clear an IP address, or a domain name is to be set here. What is not clear, is if the server is on multiple IP address whether they are on a WAN, LAN, or both, if multiple IP addresses are need, or can be added. I have a situation where I have a public WAN address, and a LAN address. Thanks a bunch!

1 0

kamailio restart:errors
by Uzma Anjum 17 Aug '17

17 Aug '17

Hi,I am new to kamailio. when i restart kamailio,I found errors on log.please help through this. 6(5633) ERROR: mymodule [my_mod.c:373]: outbound_call_process(): [outbound_call_process] URI : sip:11111111112@107.170.69.191:1008 received ip 183.82.115.246 in network ips:107.170.69.191:6090|c1|c2 6(5633) ERROR: mymodule [my_mod.c:379]: outbound_call_process(): [outbound_call_process] Request URI : 11111111112 6(5633) ERROR: mymodule [my_mod.c:395]: outbound_call_process(): outbound_call_process, obcalls privilges :select u.ob_call_privileges, a.phone_number, replace(a.name,'.','-') from account_info a, users u, subscriber s where a.id=u.account_id and u.account_id=s.account_id and u.extension=s.extension and s.username='dyan-dayan' 6(5633) ERROR: mymodule [my_mod.c:425]: outbound_call_process(): outbound_call_process query from user:dyan-dayan, phone_type:1 acc_name:dayan 6(5633) ERROR: mymodule [my_mod.c:501]: outbound_call_process(): [outbound_call_process]NEW URI : sip:ivr-1-dayan@107.170.69.191:6090 6(5633) ERROR: <core> [udp_server.c:448]: udp_rcv_loop(): recvfrom:Received encrypted packet len:4, v▒..

1 0

ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Giovanni Maruzzelli 17 Aug '17

17 Aug '17

*Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for SIP and WebRTC* Scaling your FreeSWITCH platform to serve a growing user base is a critical challenge. We'll go through the best techniques, practices, and implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, Chatting, Presence and Instant Messaging Slides can be downloaded from http://opentelecom.it/cluecon2017.pdf Enjoy ClueCon! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

4 4

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users