- sr-users - kamailio.org

call_id in dialog module
by ‪mahsa _a_‬ ‪ 20 Oct '17

20 Oct '17

hi all;i want to write "billing" project with kamailio,i using python,postgresql, dialog module and falcon.my problem is in "call_id" section.my code(this section) is in python : """... if (method == 'INVITE'): callid = req.get_param_as_list("call_id") print(req.get_param_as_list("Ts")) print("call id = %s " % callid) elif (method == 'ACK'): time_req = int(req.get_param_as_list("Ts")[0]) callid = req.get_param_as_list("call_id")[0] print("user %s call to %s at time %d % (from_user,to_user,time_req))... """ and in kamailio.cfg: """http_query("http://127.0.0.1:5000/test?method=$rm& call_id=$dlg(callid)&Ts=$Ts&fu=$fu&tu=$tu","$var(result)"); """" "callid" in ACK is ok but "callid' in invite is "null", where is my problem? thanks.

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Retransmission handling
by Volkan Oransoy 19 Oct '17

19 Oct '17

Hi all, I wonder purpose of this block in default script. What I didn’t understand is the purpose of "t_check_trans()”. Accoording to documentation, this method returns a boolean. So why do we use it even we don’t check output? # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } /Volkan

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VoLTE Capable LTE Handsets which support IMS uisng IPv4
by Ron 19 Oct '17

19 Oct '17

I would like to support VoLTE using an IPv4 IMS network rather than IPv6 network. The handsets that I have been trying insist on the IMS network support either IPv6 or IPv4v6 dual stack. After giving them IPv4v6 connectivity, I am finding that even though the handsets asked for both IPv4 and IPv6 P-CSCFs, they only try the IPv6 proxy (if only the IPv4 proxy is provided, they will not try at all). I am wondering of anyone has worked with any LTE handsets which are willing to support IMS using IPv4 only. Thanks, Ron

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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

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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

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[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>

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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

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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.

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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

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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

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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

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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

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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?

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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>

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[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 >

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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

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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)

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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

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"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 \x3c3f786d6c2076657273696f6e3d22312e302220656e636f64696e673d225554462d38223f3e3c70726573656e636520786d6c6e733d2275726e3a696574663a706172616d733a786d6c3a6e733a706964662220786d6c6e733a646d3d2275726e3a696574663a706172616d733a786d6c3a6e733a706964663a646174612d6d6f64656c2220786d6c6e733a727069643d2275726e3a696574663a706172616d733a786d6c3a6e733a706964663a727069642220786d6c6e733a633d2275726e3a696574663a706172616d733a786d6c3a6e733a706964663a63697069642220656e746974793d2232343936223e203c7475706c6520786d6c6e733d2275726e3a696574663a706172616d733a786d6c3a6e733a70696466222069643d2274384e6f3168516e7036355533336a53316f71464d677862495250712d694650223e3c7374617475733e3c62617369633e6f70656e3c2f62617369633e3c2f7374617475733e3c2f7475706c653e3c6e6f746520786d6c6e733d2275726e3a696574663a706172616d733a786d6c3a6e733a70696466223e4f6e207468652050686f6e653c2f6e6f74653e3c646d3a706572736f6e20786d6c6e733a646d3d2275726e3a696574663a706172616d733a786d6c3a6e733a706964663a646174612d6d6f64656c2220786d6c6e733a727069643d2275726e3a696574663a706172616d733a786d6c3a6e733a706964663a72706964222069643d2231223e3c727069643a616374697669746965733e3c727069643a6f6e2d7468652d70686f6e652f3e3c2f727069643a616374697669746965733e3c646d3a6e6f74653e4f6e207468652050686f6e653c2f646d3a6e6f74653e3c2f646d3a706572736f6e3e3c2f70726573656e63653e 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

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