- sr-users - kamailio.org

kamailio as application server
by eyas barhouk 01 Dec '17

01 Dec '17

Hello if i need to use kamailio as an application server like mmtel for example how could i do that and what is the features that kamailio have if i want to use it as application server in another word what is the type of application server it could be act like (mmtel - presence) ? and thanks for help

2 1

In need of a simple, stateless 302 redirect for INVITE
by Robert 01 Dec '17

01 Dec '17

Hello again, I am not a SIP expert, and I’m new to Kamailio, really enjoying learning it, but I’ve hit a blocker and I’m pretty sure I’ve missed something really simple. I’m looking to implement the Redirect Server from here: https://tools.ietf.org/html/rfc3665#section-3.6 <https://tools.ietf.org/html/rfc3665#section-3.6> (the Avaya Session Manager being Alice), Kamailio as the Redirect Server. My configuration is working fine, and I can see the 302 going back (to Avaya SM), I then get the ACK I’m expecting. The incoming SIP request contains a number of fields, including a Contact header, which I’m told SM needs on the return. But the send_reply() or sl_send_reply() doesn’t return the Contact header, and I am unable to manually set it, or any other header, using any of the append_hf(), append_to_reply() etc. mechanisms. I’m executing a msg_apply_changes() which succeeds, but no Contact header is returned. I’ve tried an sl_send_reply(“100”, “Trying”) before any other processing, and Wireshark confirms that the SIP response is basically identical to the 302 response sent after processing… The goal is to run this stateless, I’m assuming this is possible - it certainly feels like it should be. I’ve re-read the intros & tutorials, and other reading material such as this http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf <http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf> but, this, even my best Google-fu has left me utterly stumped… The sl_send_reply documentation https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply <https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply> states “If the code is in the range 300-399 (redirect reply), the current destination set is appended to the reply as Contact headers” but the Contact header is definitely not being set when I use this. Apologies in advance if eyes are rolling and this is really obvious, or (and?) that my lack of SIP knowledge is shining through - any tips or guidance would be appreciated. I certainly feel overwhelmed by the number of modules and the interplay between them. If the answers to this, and other things along these lines are contained in the admin book, that’s fab, I’ve just filled the form out :) Cheers - Robert...

3 3

DBURL password in clear
by Robert 01 Dec '17

01 Dec '17

Hello, I’m working for a UK high street bank and our Kamailio implementation has been challenged because we’ve got database passwords held in clear in the configuration file. I am unable to find any examples of where this has been worked around, there doesn’t seem to be any module or configuration means of supplying a variable in the modparam() entry that is expanded a startup. The security tutorials only seem to relate to the SIP level of security, not Kamailio as a platform. My requirement is simple, I need to be able to supply a password via means such as loading a variable from a run-once script at start up, or a module. The ideal would be to be able to read in a Docker secret :) I am by no means a Kamailio expert, so apologies in advance if this is a mindblowingly basic thing to achieve, but I do feel I’ve exhausted the Kamailio documentation, wiki etc. and all the goodness Google usually has to offer and drawn a blank. Sincere thanks in advance for any assistance. Cheers - Robert...

5 14

Dispatcher latency stats monitoring with statsd
by Julien Chavanton 01 Dec '17

01 Dec '17

Not sure if anyone is using "latency measurement" feature in Kamailio dispatcher module, I thought I could share a very simple Python script that I am using to monitor remote gateway using this. Sample Script is attached, it will post data to statsd and is working in cron The feature can be enabled : 3.27. ds_ping_latency_stats (int) Enable latency measurement when pinging nodes If set to 0, disable latency measurement. If set to 1, enable latency measurement https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc… This is one use case of this feature, another one will be to load balance based on congestion detected on latency, I am planning to share a lab experiment in a little while. Regards

2 1

[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 30 Nov '17

30 Nov '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>

3 2

blf in kamailio dmq cluster
by Aidar Kamalov 30 Nov '17

30 Nov '17

Hello, I have two(or more) kamailio servers with dmq module and dmq_usrloc - sip1.example.com and sip2.example.com publish and subcribe replication works too. BLF works fine if clients connected to the same kamailio server, because in active_watchers table stored with same domain sip1.example.com But what to do when call going from one server to second? Is there any configuration to ignore domain in pusblish, subscribe, notify, etc? -- Aydar A. Kamalov

3 7

OpenSIPS-devel not starting (TCP Error)
by SamyGo 29 Nov '17

29 Nov '17

Hi, I just installed the head version from git. version: opensips 2.4.0-dev (x86_64/linux) flags: STATS: On, SHM_EXTRA_STATS, EXTRA_DEBUG, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: d1a4419 main.c compiled on 19:53:20 Nov 28 2017 with gcc 4.8 It was compiled with EXTRA_DEBUG and DBG_TCPCON flags; While starting opensips with the residential vanilla script it gives following error: Nov 28 20:05:52 [30899] ERROR:core:tcp_init: oom con hist Nov 28 20:05:52 [30899] CRITICAL:core:main: could not initialize tcp Nov 28 20:05:52 [30899] INFO:core:cleanup: cleanup Nov 28 20:05:52 [30899] DBG:core:shm_mem_destroy: destroying the shared memory lock Nov 28 20:05:52 [30899] NOTICE:core:main: Exiting.... already running Even if I remove the proto_tcp, and tcp related things this still gives error and doesn't start. Kindly help. Regards, Sammy

2 3

parse_from_header(): bad From header
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. We experience the problem with malformed messages. First of all, I want to say, that most of time kamailio works well and nothing happens that can drop sessions. But from time-to-time something changes (may be in the network) and kamailio receives requests with malformed headers (To or From hfs). The schema: uplinks -> kamailio <-> routing server (asterisk) Malformed messages were received only from asterisk server. Sip debug on asterisk showed that messages were transmitted correctly and headers were nice. But kamailio obtains changed (broken) packets or perhaps can't read them properly. How does it look like: Nov 10 12:37:06 kamailio-name kamailio[965]: INFO: <script>: Going to NATMANAGE for BYE from sip:useragent7122@kamailio_address:5068 (IP=wss:client_address:62881) - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: NOTICE: <script>: Relaying request to <null> - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: uac [replace.c:783]: restore_uris_reply(): failed to find/parse FROM hdr Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: Skip manage DEVICE_STATE for BYE from <null> (IP=udp:asterisk_address:50600) with (200 - OK) - R=<null> ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: ------------------------------------MANAGE BYE by DEVICE_STATE_BYE_MANAGE route - 200 - OK M=BYE IP=udp:asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine_funcs.c:331]: get_from_tag(): failed to parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine.c:2252]: rtpp_function_call(): can't get From tag Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header How the packet looks like: ACK sip:tt7etmau@bb0pd5t63b26.invalid;alias=client_address~57573~6;transport=ws SIP/2.0 Via: SIP/2.0/UDP asterisk_address:50600;branch=z9hG4bK117106ca;rport Route: <sip:kam2.domain.com:5068 ;nat=yes;transport=udp;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331>,<sip:kam2.callision.com:5068 ;nat=yes;transport=ws;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331> Max-Forwards: 70 From: "19172423539" <sip:19172423539@asterisk_address:50600>;tag=as06869a5d To: <sip:useragent7122@10.0.1.18:5068>;tag=nmv7kmpau3 Contact: <sip:19172423539@asterisk_address:50600> Call-ID: 629d5a132c194c536b5f5c1a2a3c6e32@asterisk_address:50600 CSeq: 102 ACK User-Agent: asterisk PBX Content-Length: 0 The transport is used to send messages between asterisk and kamailio is TCP. TCP configurations: tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 I found the idea, that we need to change following parameters: modparam("uac","restore_mode","auto") modparam("uac","restore_dlg",1) to: modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) I did that, now it looks like all is fine, but I think it can get back. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua>

3 5

Source socket/port for in-dialog requests
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. My apologies for so frequent appealing to you. I'm trying to solve problem with ending of sessions. The problem consists of no 200 OK coming from uplink on our BYE requests. Topology. First leg: Webrtc client <-wss-> kamailio <-sip tcp-> asterisk routing server Second leg: Uplink <-sip tcp-> kamailio <-sip tcp-> asterisk routing server The problem appears only in case when dialog was ended by webrtc client. The first leg (dialog) of the call ends nice, without any hints on problem. But the second leg (with uplink) has problem with no 200 OK (coming from uplink) on BYE request coming from asterisk. Handshake between asterisk server and uplink establishes properly. It looks like: uplink.domain.com:5060 -> INVITE tcp -> our.kamailio.server:5060 -> INVITE tcp -> our.asterisk.server:5060 . uplink.domain.com:5060 <- 100 trying tcp <- our.kamailio.server:5060 . uplink.domain.com:5060 <- 180 ringing tcp <- our.kamailio.server:5060 <- 180 ringing tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 <- 200 OK tcp <- our.kamailio.server:5060 <- 200 OK tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 -> ACK tcp -> our.kamailio.server:5060 -> ACK tcp -> our.asterisk.server:5060 . uplink.domain.com <-media stream-> rtpengine <-media stream-> our.asterisk.server . Here kamailio starts to use random source port for relaying in-dialog BYE uplink.domain.com:5060 <- BYE tcp <- our.kamailio.server:45355 <- BYE tcp <- our.asterisk.server:5060 . And here the leg with uplink is expected to end with 200 OK (coming from uplink proxy). But uplink doesn't answer at all. We requested a tcpdump from uplink to see how packets are forwared from their side. And I saw that that 200 OK tries to be sent within first tcp session - to 5060 port of kamailio server. But sngrep on our side shows nothing, nothing appears in kamailio log, so 200 OK can't reach the 5060 socket because of some transport problem I think. Top via hf of our BYE request, contains kamailio record = uplink.domain.com:5060 My main thought on this count is to suppress using of random source ports for in-dialog requests, this behaviour looks irrelevant. We have turned on mhomed parameter (mhomed=1). And cookbook says: "Set the server to try to locate outbound interface on multihomed host. This parameter affects the selection of the outgoing socket for forwarding requests." But, there is a big but - we can not turn of this parameter for this moment, because routing script made with using of this one. Kamilio works on AWS EC2 platform. It has following IP address schema: private address atouched to container -> public address assinged to this private address (so this is standard ip address schema for AWS containers). For this example private address will be: 10.0.0.1 and public address will be: 100.0.0.1 Configuration - main parameters (all values are changed for an example): advertised_address=our.kamailio.server advertised_port=5060 alias=our.kamailio.server alias=our.kamailio.server:5060 alias=100.0.0.1 alias=100.0.0.1:5060 alias=10.0.0.1 alias=10.0.0.1:5060 auto_aliases=no port=5060 listen=100.0.0.1 listen=10.0.0.1 listen=tcp:100.0.0.1:5060 listen=tcp:10.0.0.1:5060 mhomed=1 fork=yes fork_delay=5000 children=6 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 One of the solutions to this problem was to add following inside the relaying route block: $fs = "tcp:" + "10.0.0.1" + ":5060"; if (!t_relay()) { sl_reply_error(); } But it seems to me it's not a smart solution. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 1

Best way to do routing against a list of gateways
by voipspace voipspace 29 Nov '17

29 Nov '17

3 6

allow_trusted not passing
by voipspace voipspace 29 Nov '17

29 Nov '17

4 5

URL problems connecting to external (Azure) MySQL database
by Floimair Florian 29 Nov '17

29 Nov '17

Hi! I'm having troubles connecting Kamailio to an external MySQL database (on Microsoft Azure Cloud). Initially creating the database with kamctl worked fine, but accessing it at runtime via db_url fails. A somewhat unusual thing about the connection to a MySQL database on Azure is that the username contains an '@' character. This leads me to believe that this case is somewhat unhandled when parsing the database URI. Trying to connect to something like modparam("sipcapture", "db_url", "mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db") Results in errors starting Kamailio ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'clouddb' (-2) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool ERROR: sipcapture [sipcapture.c:1146]: child_init(): unable to connect to database [mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db] from capture_mode param. Is there a trick on how to achieve a connection in this case like some sort of escape character probably? Mit freundlichen Grüßen Florian Floimair (IMS) 1568

2 4

lcr_id vs lcr_count
by Patrick Wakano 29 Nov '17

29 Nov '17

Hello Kamailio list! I've been working with the LCR module and I wanted to use the lcr_id to match some ID's I have somewhere else. However, due to the lcr_count my lcr_ids must be less than the lcr_count. I understand the reason behind the lcr_count, what I think is a mistake is to have the lcr_id dependent of the lcr_count. Ideally we should be able to use any lcr_id value, just observing that the total amount of lcr groups should not exceed the lcr_count. The problem is this if that appears in the load_gws, to_gw and from_gw functions in lcr_mod.c: if ((lcr_id < 1) || (lcr_id > lcr_count_param)) { LM_ERR("invalid lcr_id parameter value %d\n", lcr_id); return -1; } I ended up using another table to make the conversion of ids, but this is horrible.... Any other reason to force the lcr_id to be less than the lcr_count? Best Regards, Patrick Wakano

3 3

how to add new media server in dispatcher table through "ANSIBLE"
by Narayan P 29 Nov '17

29 Nov '17

Hello Experts, I am new to ansible and deploying kamailio through ansible.Sucessfully deployed kamailio but my requirement is to add media server through ansible.As I am using dispatcher module in kamailio,not able to find any script that directly insert one media server IP and port into dispatcher table from ansible.(I am using ansible in one system and deploying kamailio in other). Any suggestion will be highly appreciated as always. Thanks, Narayan +91-9035438928

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Re: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue.
by otron2016＠gmail.com 29 Nov '17

29 Nov '17

Through Firefox likely means via websockets (wss) rather than via the sip port. Many things could interfere with the latter. Take heart, though: websockets is harder to set up so you're likely through the worst. Look at logs and see what iftop shows you about traffic on the server as you try to make sip connections. Also try your sip client with a known-good sip server (e.g. Iptel) just to make sure the problem isn't with the client. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/28/2017 6:39 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue. Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won’t work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don’t know 100% for sure. Thank you

1 0

Using SIP tools through Kamailio and Asterisk - chrome issue.
by Wilkins, Steve 28 Nov '17

28 Nov '17

Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won't work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don't know 100% for sure. Thank you

1 0

dmq replicate
by Aidar Kamalov 28 Nov '17

28 Nov '17

Hello, I set up kamailio cluster. dmq_usrloc works fine, but PUSBLISH and SUBSCRIBE replication not. If I make a call to first node I've got many(hundreds) PUBLISH messages from first node to first node and some messages (8 for example) to other nodes. And second xnotice in config not executes. And errors n log ERROR: <core> [core/sr_module.c:1726]: get_int_fparam(): Unsupported param type for int value: 32 How to replicate publish and subscribe, is it wrong way? my confiig: request_route { if(is_method("KDMQ")) { if ($Rp == 5050) { dmq_handle_message(); exit; } else { drop; } } ..... ..... if(is_method("PUBLISH|SUBSCRIBE")) { if(!dmq_is_from_node()) { xnotice("AAAAAAAA"); dmq_t_replicate("1"); xnotice("AAAAAAAABBBBBBBBBBB"); } } 2017/11/28 12:56:06.415601 192.168.8.213:5060 -> 192.168.8.213:5060 PUBLISH sip:34200@siptest. SIP/2.0 Via: SIP/2.0/UDP 192.168.8.213;branch=z9hG4bKc7a5.cee84051000000000000000000000000.0 To: <sip:34200@siptest> From: <sip:34200@siptest.>;tag=ae81df16b5a52fc1f62f9b45f337e660-db10 CSeq: 10 PUBLISH Call-ID: 4c8f9e6e799efcd7-57244(a)192.168.8.213 Content-Length: 556 User-Agent: kamailio (5.0.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Expires: 125 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:34200@siptest."> <dialog id="898932197-5060-86(a)BJC.BGI.D.DB" call-id="898932197-5060-86(a)BJC.BGI.D.DB" direction="initiator"> <state>Trying</state> <remote> <identity>sip:53003@siptest.t</identity> <target uri="sip:53003@siptest.tg"/> </remote> <local> <identity>sip:34200@siptest</identity> <target uri="sip:34200@siptest"/> </local> </dialog> </dialog-info> -- Aydar A. Kamalov

1 1

dispatcher/carrierroute and branches across SBC and remote peer
by Nathan Ward 28 Nov '17

28 Nov '17

Hi, We use Kamailio as a SIP server, and SEMS as an SBC. We have a need to use the dispatcher/carrierroute modules to provide HA/load sharing across two different parts of our environment: 1) Our SBCs 2) Our peer’s SIP servers We have pairs of SBCs (for now) and they have the same network connectivity - different IPs of course, but they can reach the same things. Our peer’s network has 2 SIP servers (again, for now) reachable through either SBC. We select which SBC pair to use based on where the calls are going, in a branch route. In some cases, we use parallel forking, in others we use serial forking (i.e. to a media server for a prompt, then sending the call to wherever it needs to go). Right now there is no automatic failover or load sharing, so this is all simple logic - if call is to x peer or y customer, use x SBC or y SBC. I’m trying to add the failover/loadbalancing now. I have a few half baked ideas about how to achieve a network like this, but thought I’d ask if anyone else had done this first before I went too overboard. Currently I’m thinking of calling cr_route to figure out which peer SIP server, from there figuring out which SBC pair, and then calling ds_select_dst to select which SBC in the pair to set $du. I’m being tripped up though, as I can’t call cr_route or ds_select_dst in a branch route, but, as far as I’m aware that’s where I’d need to do so in order to handle this for multiple branches. Should I be looping the request through the SIP server to achieve this (i.e. force $du = 127.0.0.1 in my branch_route), or, can I avoid that complexity? -- Nathan Ward

1 0

Kamailio didn't start before increasing fork_delay
by igor.potjevlesch＠gmail.com 27 Nov '17

27 Nov '17

Hello, We suddenly had an issue on one Kamailio instance: we were not able to restart. Kamailio started to boot and fork and suddenly crashed. The last logs reported a failure regarding the ability to connect to one MySQL instance. I finally succeed to restart after increasing: fork_delay=5000 to fork_delay=9000. How this could happen suddenly? We already restarted Kamailio on this server. Regards, Igor.

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Does Kamailio support RFC4117?
by Ye, Charlie (NSB - CN/Qingdao) 27 Nov '17

27 Nov '17

Hi Friends, Does Kamailio support RFC4117? Do you have any suggestions if I want support RFC4117 through Kamailio? Thanks and Best Regards, Charlie

2 2

Dispatcher with kam & fs behind nat
by Daniel Greenwald 27 Nov '17

27 Nov '17

Dispatcher is sending inbound calls to Freeswitch boxes. Both the FS boxes and Kam are behind the same nat. What is the correct way to pass calls to the FS boxes so that all SIP between FS & Kam use local IP's? Should Kam put two record-route headers on, one with PUBLICIP and one with PRIVATEIP? How is this accomplished? Thanks

2 3

Kamailio 5.0 Android Calling Not working
by Zeeshan M 25 Nov '17

25 Nov '17

Hello Friends, I had successfully installed and configured Kamailio 5.0 on linux machine. And tried it with linphone app & following are the issues i am facing, 1) IOS to Android Calling not working with timeout error on server end when we checked logs 2) Android to Android calls also not working same error. Kindly let me know what could be the issue. Regards, Zeeshan.

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Re: [SR-Users] sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

I just verified that I do have logging in all pertinent areas. I added tls and I do see "TLS accept successful" in the kamailio logs, however, I see no log being generated in route[xhttp:request], and I do have logging in the first line of this route. I don't think it is getting there. Also, I noticed I have event_route[xhttp:request] not route[xhttp:request]. Thank you, -Steve

1 0

sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen's in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve

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Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

Just a guess but maybe later entries [like +HIGH:+MEDIUM:+LOW] put it back. Try switching the order so that !ECDHE and the others you're trying to exclude come after. Sent from Samsung Mobile -------- Original message -------- From: Ilyas Keskin <ilyask92(a)gmx.de> Date: 11/24/2017 10:19 AM (GMT-08:00) To: miconda(a)gmail.com,"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite Hi Daniel, yes I am using the tls.cfg file. I tried your suggestion to add the cipher suite string (notice the !EDCHE which I also added to the httpd ssl.conf) but nothing changed. [server:default] method = TLSv1 cipher_list = !DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem certificate = /etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem #ca_list = ./modules/tls/cacert.pem #crl = ./modules/tls/crl.pem Also here is a log snippet from tls module section of kamailio initialization. Notice first two lines. Also it seems to me the module actually ignores the local openssl installation and uses its own which has been compiled with the module itself (?). Other than that it seems to be accepting the cipher_list value just fine: Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:355]: mod_init(): With ECDH-Support! Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:358]: mod_init(): With Diffie Hellman Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:587]: init_tls_h(): tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, compression: on Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:595]: init_tls_h(): tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, zlib compression: compiler: gcc -I. -I.. -I../include -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_ Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tls [tls_init.c:649]: init_tls_h(): tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 7864320 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 3932160 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_branch("MANAGE_BRANCH"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_reply("MANAGE_REPLY"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_failure("MANAGE_FAILURE"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSs<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSs<default>: certificate='/etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSs<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSs<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSs<default>: require_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSs<default>: cipher_list='!DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSs<default>: private_key='/etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSs<default>: verify_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSs<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:670]: set_verification(): TLSs<default>: No client certificate required and no checks performed Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSc<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSc<default>: certificate='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSc<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSc<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSc<default>: require_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSc<default>: cipher_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSc<default>: private_key='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSc<default>: verify_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSc<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:655]: set_verification(): TLSc<default>: Server MUST present valid certificate Would it be possible to compile the tls module with certain openssl config switches (i.e. no-ec no-dh)? Any other ideas? Best regards, Ilyas Keskin Am 24.11.2017 um 15:45 schrieb Daniel-Constantin Mierla: Hello, On 23.11.17 22:42, Ilyas Keskin wrote: Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. are you also using tls.cfg? If yes, there is an attribute for chiper list in it as well, try and see if works with it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Re: [SR-Users] sipML5 through kamailio
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

"So, tls is required for wss?" Yes, as of the last rfc. It can work without it if the client software (browser) will accept it though by now most will treat it as insecure. I forget whether sipML5 will but my guess would be no. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/24/2017 9:57 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hello SamyGo, So, tls is required for wss? I thought that when I set a listen to Kamailio-Server-IPAddress:10443 and I sent a request through that IP and Port, that Kamailio would at least see the request and attempt to do something with it. I will go add tls and try again. Thank you! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Can you clarify the IP addressing scheme as you've mentioned. There is no TLS interface? Advertised address for 5060 but not for WSS interface. Both listen and advertise are public IPs ? Also,you're handling WSS requests. Do you've xlog ines in the route[xhttp:request] to view when a request lands.! My config has this: #!substdef "!MY_IP_ADDR!123.134.156.167!g" #!substdef "!MY_WS_PORT!6010!g" #!substdef "!MY_WSS_PORT!6011!g" listen=tcp:MY_IP_ADDR:MY_WS_PORT listen=tls:MY_IP_ADDR:MY_WSS_PORT Then the xhttp event route: ``` event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); #Deny any HTTP requests on any port other than WS/WSS ports. if ($Rp != MY_WS_PORT && $Rp != MY_WSS_PORT ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } #Handle HTTP(s) onwards. xlog("L_INFO", "HTTP Request Received\n"); ``` On Thu, Nov 23, 2017 at 12:18 PM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hi Sammy, First of all, thank you for taking the time to respond. Yes, port 10443 is opened. I have used this port before as asterisk’s WebRTC port and iptables shows it as open. No, I can’t even get a registration using the configuration I listed. I have an xdbg log statement right after the request_route, and I see nothing. I do know that my xdbg logs are working though because, if I register or make a call using any sip tool, I see all my logging and everything works correctly. -Steve From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:00 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hi Steve, Can you confirm that port 10443 is reachable behind the NAT to Kamailio server, validate iptables too Does your SIPml5 demo client register successfully to Kamailio? are there enough xlog lines to print out if anything lands in Kamailio. Regards, Sammy On Thu, Nov 23, 2017 at 11:34 AM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen’s in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve _______________________________________________ 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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Cannot disable EC Diffie Hellman cipher suite
by Ilyas Keskin 24 Nov '17

24 Nov '17

Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. Best regards, Ilyas Keskin

2 2

Kamailio KEMI Framework Tutorial
by Daniel-Constantin Mierla 24 Nov '17

24 Nov '17

Hello, I started a tutorial for Kamailio KEMI Framework written in markdown format targeting use of mkdocs. It is published on github as part of kamailio-docs repository: * https://github.com/kamailio/kamailio-docs/tree/master/kamailio-kemi-framewo… The mkdocs build (to browse and read) is online at: * http://kamailio.org/docs/tutorials/devel/kamailio-kemi-framework/ It is work in progress and contributions from the community are more than welcome -- doing push requests on github project is the recommended way for contributions: * https://github.com/kamailio/kamailio-docs/ The main goal is document the role of KEMI, its benefits and how it can be used. It addition it should cover the list of functions exported to KEMI by core as well as special KEMI functions. Regarding the functions exported by kamailio modules to KEMI I added as an skeleton the ones exported by acc module. However, I am not sure if we should do it there or add them to the existing docs of the modules. Opinions on this?!? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

2 10

Request Time Out
by Vasim Antule 24 Nov '17

24 Nov '17

Hello, I am new to SIP/Kamailio , I am attempting to use Kamailio 5.0 in test environment. I am getting Request Time Out for Android Phones. Apple to Apple phones are working fine. But Android to Apple and Android to Android I am getting Request Time Out. Please Guide. Regards, Vasim Antule

1 0

Issue dumping an huge htable through xmlrpc
by Marco Capetta 24 Nov '17

24 Nov '17

Hello everyone, On Kamailio version 4.4.6 I have an issue dumping an huge htable through xmlrpc. We are using htables to store the subscriber registration attempts to prevent bruteforcing password attacks. Usually there aren't any issues to dump information using xmlrpc. On the contrary, if the number of records in the htable become huge (in the order of 10.000), xmlrpc output is truncated, therefore it is not possible to parse it. I have the same results either using a Perl library or the "xmlrpc_test.py" script included in package. I tried to increase the module parameters "binrpc_max_body_size" and "binrpc_struct_max_body_size" without success. Do you have any other suggestion? Thank you Regards Marco -- *Marco Capetta * Operations Engineer Sipwise GmbH <http://www.sipwise.com> , Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Phone: +43(0)1 301 2044 <tel:+4313012044> Email: mcapetta(a)sipwise.com <mailto:mcapetta@sipwise.com> Website: www.sipwise.com <http://www.sipwise.com> Particulars according Austrian Companies Code paragraph 14 "Sipwise GmbH" - Europaring F15 - 2345 Brunn am Gebirge FN:305595f, Commercial Court Vienna, ATU64002206

2 1

Failed to find command http_client_query
by Alexandru Covalschi 24 Nov '17

24 Nov '17

Hello everyone, Updated to 5.0.4 from repo, up-to-date debian 8, http_client.so is certainly loaded. When I try to change http_query to http_client_query it fails to load with error "failed to find command http_client_query". Any ideas? It is same on all my test servers. ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493

2 12

msilo support for picture, audio and video?
by sscc 24 Nov '17

24 Nov '17

Hello Folks does msilo support picture, audio and video? regards sscc? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 5

Loadbalancing authenticated agetways
by SamyGo 23 Nov '17

23 Nov '17

Hi, I've a scenario where multiple mediaservers will be registered to Kamailio and for any incoming call from Upstream provider I've to perform load-balancing to the actively registered media-servers. dispatcher module does load-balancing for IP endpoints, is there any other module where I can perform load-balancing on registered clients(media-servers)? Thanks, Sammy

3 3

why do I need rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk via Kamailio, and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Thank you

3 2

rfc: clang format the code
by Daniel-Constantin Mierla 23 Nov '17

23 Nov '17

Hello, it was discussed during the last irc devel meeting and everyone there agreed to use clang-format to format the source code: - https://clang.llvm.org/docs/ClangFormat.html Is any developer (maybe not present during the irc devel meeting) opposing this? It is not going to be all at once, because some of the structures and initialization of them may be preferred to be kept with a custom format for being easier to read -- to have a section in the file not formatted, it has to be enclosed in between comments: /* clang-format off */ ... /* clang-format o. */ But for the rest of the code (i.e., functions) I think it will be better to agree on a format and try to use it. The clang-formatting should be done when one needs to update code in some files. Over the years, developers come and go, having each one with a custom formatting style is hard to maintain properly in long term. There are modules that do not indent coherently code inside blocks, using a mixture, more or less no style. This makes it hard for other developers to review in case of reported issues, but also for new comers to get familiar with the code. In the source tree there is already a .clang-format file which defines the options for formatting. It was used by me and other developers already -- I did it mainly in my modules or those components that do not have an active maintainer. But I plan to do it also on the components that have currently a maintainer at the moment I need to work on that code (e.g., fix some issue or export functions to kemi framework). My process of clang-formatting a module is like: - edit the files to add clang-format off/on comments around the parts I want to preserve as they are (typically mod exports -- params, functions, kemi -- and intialization of structures that are aligned to a specific indentation that makes it easier to read). - run: clang-format -i src/modules/modname/*.{c,h} - commit with these changes, no new feature or bug fix must be done in the same commit (can be combined with removing the history comments) If one wants different options inside .clang-format, please propose and let's discuss which one is preferred by the majority of the current developers. If no one shows anything against in a matter of few days, it will be considered accepted (after all, one can re-indent on own style if the rule is not accepted or overturned). Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Error when attempting to use rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

Hello, I entered this question before but did not include the tcpdump file. I am re-posting just in case this message was over looked for lack of information. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. My server name is vadev2 (Kamailio run on this server), and it's IP Address and port are 172.21.1.108:5060; Asterisk servers are (172.21.1.97:5060, and 172.21.1.124:5060). The rtpproxy is 172.21.1.108:17722 Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 vadev2/usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.21.1.108:59794<http://172.1.1.13:59794> failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work, other than I do not believe anything is going through rtpproxy; I can make calls and use sip tools which pass requests through Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what and why. Should I be using rtpengine instead of rtpproxy? I did attempt rtpengine but it would not compile due to missing json.h files. Thank you for any help -Steve

1 0

MSILO with encryption/decryption
by SamyGo 23 Nov '17

23 Nov '17

Hi, Is there any internal mechanism for msilo to store messages encrypted and retrieve them decrypted before sending? Thanks, Sammy

1 0

RV: CANCEL not being matched by tm module when via address does not have port included
by CARLOS GARCIA MORCHON 23 Nov '17

23 Nov '17

Hi, I have an IMS core deployed using kamailio 5.0.2. Calls from clients are reaching the core pcscf from a freeswitch (with address A.A.A.A) + a kamailio acting as proxy (with address B.B.B.B), so the INVITE request looks like this: INVITE sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Record-Route: <sip:A.A.A.A;lr> Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Via: SIP/2.0/UDP B.B.B.B:5080;received=B.B.B.B;rport=5080;branch=z9hG4bKHpB704ZaXSKtm Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 INVITE Contact: <sip:mod_sofia@B.B.B.B:5080> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 224 P-Charge-Info: <sip:222222222@mobile.tuenti.int>;npi=ISDN;noa=4 X-FS-Support: update_display,send_info Remote-Party-ID: "222222222" <sip:222222222@B.B.B.B>;party=calling;screen=yes;privacy=off If the client cancels the request, the CANCEL request that gets to the core pcscf looks like this: CANCEL sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 CANCEL Content-Length: 0 Reason: Q.850;cause=16;text="Normal Call Clearing" The processing of the CANCEL at the pcscf is eventually reaching the t_check_trans() function, but the logs return “no CANCEL matching found!” and thus the processing ends. I have followed the code up to the via_matching function in the t_lookup.c file of the tm module source and added some logs. The via matching is failing at the port comparison (the port stored for the INVITE is “5060”, the port stored for the CANCEL is “0”. The flow works ok with a different client (connected directly to the core) which includes the port in the Via headers. ¿Can this be a bug in kamailio or am I doing something wrong?. AFAIK The RFC 3261 does not mandate to include the ports in the sent-by element of the Via header Thanks! Carlos ________________________________ Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it. Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

3 4

Kamailio new install, unable to create users
by Atux Atux 23 Nov '17

23 Nov '17

Hi. New to the area and trying to find my way around with kamailio. In one of my debian servers i need to have kamailio with a gui to start playing around with it. i am following the http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour to install kamailio. When running kamdbctl create it asks what format coding that i must have to configure the DB. i simply add latin1 and it goes on. then when i issue kamctl add daniel 1234qwet it comes with the following error: root@debian:~# kamctl add daniel 1234qwet MySQL password for user 'kamailio@localhost': ERROR 1045 (28000): Access denied for user 'kamailio'@'localhost' (using password: YES) ERROR: introducing the new user 'daniel' to the database failed root@debian:~# Any ideas on how to proceed with the following error please?

2 3

kemi with lua and dispatcher - attempt to index field 'dispatcher' (a nil value)
by Karsten Horsmann 23 Nov '17

23 Nov '17

Hello Daniel, hello List, after reading the fresh released kemi documentation (Thanks for that!) on github i start to configure an kemi lua configuration. It seems to be that there is not an dispatcher export to kemi in my installation. Or i missconfigured something? How to use dispatcher with kemi? 1.) First try to solve this - use param "register" "modulename" to app_lua - doesnt work 2.) grep around in the auto kemi in/export messages in debug startup - nothing to see. 3.) grep around in the kamcmd api.list command for lua - no dispatcher here the intressting parts of the configs and lua-script: kamailio.cfg ... loadmodule "dispatcher.so" loadmodule "permissions.so" loadmodule "avpops.so" #!ifdef WITH_CFGLUA loadmodule "app_lua.so" modparam("app_lua", "reload", 1) modparam("app_lua", "register", "dispatcher") #!endif #!ifdef WITH_CFGLUA modparam("app_lua", "load", "/etc/kamailio/lua/kamailio-kemi-lua.lua") cfgengine "lua" #!else import_file "request-route.cfg" #!endif ... kamailio-kemi-lua.lua function ksr_request_route() ... -- dispatch destinations if KSR.permissions.allow_source_address(FLT_EDGE) then KSR.info("new $fU -> $rU in permissions"); direction = "inbound"; routing_target = FLT_FSNODE; ksr_route_dispatch(direction, routing_target); KSR.x.exit(); end KSR.info("NOT-OKAY - $rU dropped"); -- End of Route() KSR.sl.sl_send_reply(407, "Proxy Authentication Required. See IP GUI"); return 1; ... end --- dispatcher route function ksr_route_dispatch(direction, routing_target) KSR.xlog.xinfo("new $fU -> $rU in routing_target ksr_route_dispatch"); KSR.info(routing_target); if (routing_target == nil) then KSR.xlog.xinfo("routing_target nil"); KSR.sl.sl_send_reply(604, "Does Not Exist Anywhere"); KSR.x.exit(); end if KSR.dispatcher.ds_select_dst(routing_target, 4) then send_reply(404, "No destination"); KSR.x.exit(); end ksr_route_myheader(); ksr_route_relay(); KSR.x.exit(); end /usr/sbin/kamailio[26763]: ERROR: app_lua [app_lua_api.c:713]: app_lua_run_ex(): error from Lua: /etc/kamailio/lua/kamailio-kemi-lua.lua:131: attempt to index field 'dispatcher' (a nil value) kamailio -v version: kamailio 5.0.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 20:20:24 Nov 16 2017 with gcc 4.8.5 rpm -aq | grep kam kamailio-mysql-5.0.4-26.el7.centos.x86_64 kamailio-5.0.4-26.el7.centos.x86_64 kamailio-lua-5.0.4-26.el7.centos.x86_64 kamailio-ims-5.0.4-26.el7.centos.x86_64 rpm -ql kamailio-5.0.4 | grep disp /usr/lib64/kamailio/modules/dispatcher.so /usr/share/doc/kamailio/modules/README.dispatcher /usr/share/kamailio/dbtext/kamailio/dispatcher -- Kind Regards *Karsten Horsmann*

2 5

"Federated" PBX solution with GUI to use with Kamailio - Suggestions needed
by George Diamantopoulos 23 Nov '17

23 Nov '17

Hello all! I'm looking at "PBX" platforms to use with kamailio as a load-balancer. This is not a strictly kamailio question, but I thought that if anyone has a suggestion to make, they would be watching this list. I'm investigating a use case with the following requirements: * Kamailio will act as a load-balancer, dispatching initial requests to PBX entities (e.g. Asterisk, Freeswitch) * PBX solution must have a single GUI interface to manage PBX functions (think FreePBX, FusionPBX etc) * Configuration via the GUI must apply to all PBX instances * Some sort of "asterisk-realtime + device state synchronization"-like setup must ensure every PBX box can service any request at any time (for failover) * Solution needn't be multi-tenant oriented, but if it is, it must work fine for single-tenant setups as well without multi-tenant capability getting in the way all the time * Feature list needn't be as full as, say, FreePBX's, but most common functionality must be available (inbound/outbound routing, trunks, queues, hunting to name a few) * Must be open source As a proprietary example, MiRTA PBX seems to check all the boxes above, apart from being open source. Now this "product" also offers high availability for databases and web apps, which is beyond the scope of this investigation. I'm only looking at a solution that provides a federated, synchronized farm of PBX instances. Any input would be greatly appreciated. Thanks! Best regards, George

3 2

Re: [SR-Users] Kamailio IMS + Asterisk PBX
by Amar Tinawi 22 Nov '17

22 Nov '17

My tests results : Unfortunately, no scenario was success, maybe because Asterisk does not support sh interface (Diameter) My test was for presence service and couldn’t get it done. The results of Having asterisk as an AS with kamailio IMS : with sip trunk configured towards S-cscf And whether the user is identified in Asterisk as extension user or not asterisk responded with 404 user not found. Sent from Mail for Windows 10

2 1

debian setup without registration
by Atux Atux 22 Nov '17

22 Nov '17

Hi. i am new to the are of kamailio. i am looking to install in a PC running debian 9 kamailio. that is easy since all i had to do was apt-get install kamailio. I am looking on how to setup the system in a way that everyone sending a register, to be able to register no matter what user name or passwd sends. i do not want a MYSQL server on that server. Could someone give me some hints on how to configure it please? In /etc/kamailio/kamctrlc i have SIP_DOMAIN=192.168.124.5 which is my IP. Everybody from my LAN 192.168.124.0/24 network must be able to send a register with any username and any passwd and be able to register. Also from the outside world, through port forward from my router. Some hints please?

3 2

kamcmd gives segmentation fault (5.0.4)
by José Lopes 22 Nov '17

22 Nov '17

Hello, I invoke kamcmd inside Freeswitch Lua Scripts and right now i am getting segmentation fault any time. This started to happen after I upgraded kamailio from 4.4 to 5.0.4 . I tried to analyse the core dump and it give me the next information. When I execute on the console, the command works. Am I doing something wrong? There is any alternative or better way to communicate with kamailio? Thanks in advance Best regards, Jose Lopes Core was generated by `/usr/sbin/kamcmd -s udp:127.0.0.1:2050 uac.reg_refresh s:29010017'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 591 kamcmd.c: No such file or directory. (gdb) bt #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 #1 0x0000000000403eee in run_binrpc_cmd (s=3, cmd=0x7ffec898a8d0, fmt=0x0) at kamcmd.c:924 #2 0x0000000000404499 in run_cmd (s=3, cmd=0x7ffec898a8d0, format=0x0) at kamcmd.c:1068 #3 0x00000000004073aa in main (argc=5, argv=0x7ffec898d1e8) at kamcmd.c:2114

2 2

Kamailio IMS + Asterisk PBX
by Amar Tinawi 21 Nov '17

21 Nov '17

Dears; Good day ! is it possible to use Asterisk pbx server as an Application server for the kamailio IMS users ? as if i want to correlate an IMS user with an extension number. My though is kamailio ims will handle the registration process and authentication and Asterisk will deal with IMS users as they are PBX users to get all asterisk's benefits (call conference, monitoring, waiting, voice mail ... etc) i'm thinking to integrate Asterisk with S-cscf, and when an IMS user register to the IMS to domain, a trigger will initiate the S-cscf to send notify to asterisk . i didnt find any solution or tutorial for this procedure, and idont know if it is possible to be done. Regards Ammar

5 8

Adding strip and preix attribute to Dispatcher table
by Ashutosh chaubey 21 Nov '17

21 Nov '17

Hi, Is there a way to add prefix and strip in dispatcher table attribute column. I have tried prefix=11,strip=3 but this seems to not work. Any help? Regards, Ashutosh Chaubey

2 4

http_async_client problem with tls_ca_path
by gmele 20 Nov '17

20 Nov '17

Hello, I'm having strange behavior with the http_async_client module and https connections. Our kamailio config use REST interfaces to send push to mobile apps when calls are initiated. We also use a REST interface to register the client in our push database (and then be able to use the info to send a push). Our REST interface use TLS to cypher exchanged data between the SIP registrar and the Pusher service. We have configured the tls connection using modparam: # ------ http async client params ------- modparam("http_async_client", "workers", 16) modparam("http_async_client", "connection_timeout", 20000) # 20 seconds timeout on HTTP requests modparam("http_async_client", "hash_size", 2048) # update this value depending of the load #modparam("http_async_client", "curl_verbose", 1) # curl debug modparam("http_async_client", "tls_ca_path", "/etc/kamailio/ssl/ca") What we have noticed is when there is 1 SIP Register requests triggering and HTTPS request all works fine, but as soon as we have several HTTPS request in //, the HTTPS request fails with curl error 77: curl: (77) Problem with the SSL CA cert (path? access rights?). We tried to put the Root CA Path when building the request: $http_req(all) = $null; # reset the parameters $http_req(method) = "POST"; $http_req(hdr) = "authorization: "+$hdr(authorization); $http_req(hdr) = "Content-Type: application/json"; $http_req(suspend) = 0; # don't suspend the transaction, continue routing script's execution $http_req(body) = "{ 'pushRegistration':{'callId':'" + $ci + "', 'fromUri':'" + $fU + "@" + $fd + "', 'contactLine':'" + $ct + "'}}"; $http_req(tls_ca_path) = "/etc/kamailio/ssl/ca/"; http_async_query("https://pusher:443/rest/push/register", "HTTP_REGISTER_REPLY"); But we have exactly the same behavior. Does the http_async_client module supports muti-thread? Regards Giovanni -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 17

Re: [SR-Users] 183 acc records even if early_media equals to 0
by Daniel-Constantin Mierla 20 Nov '17

20 Nov '17

Hello, can you run list command in gdb to see the code? Because I am not sure what is exactly at that line in your version with the patch. Cheers, Daniel On 09.11.17 18:10, Marco Capetta wrote: > I applied the patch but kamailio crashes with the following error: > > (gdb) bt full > #0 0x00007f8af1e9b9ae in relay_reply (t=t@entry=0x7f8a29c05868, > p_msg=p_msg@entry=0xffffffffffffffff, branch=branch@entry=0, > msg_status=<optimized out>, > cancel_data=cancel_data@entry=0x7ffd92443570, > do_put_on_wait=do_put_on_wait@entry=0) at t_reply.c:1968 > relay = 0 > [...] -- 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

3 7

Dmq usage
by Aidar Kamalov 20 Nov '17

20 Nov '17

Hello, I want to distribute some messages-actions for redis to my kamailio servers. I use this code: For send: dmq_bcast_message("redis", "set early:$fU", "text/plain"); For recieve: request_route { xnotice("ROOT: $rm $rU [$ci]"); if(is_method("KDMQ")) { if($rU == 'redis') { route(REDIS); } else { dmq_handle_message(); } } ...... route[REDIS] { xnotice("REDIS: $rb"); redis_cmd("redis", "$rb", "r"); return; } When dmq_bcast_message starts - other node recieve 7 KDMQ sip packets. And not always it executed. Is it right way to use dmq? Dmq user locations are ok.

2 8

kamailio-wiki dead links to github.com/....master branch
by Karsten Horsmann 20 Nov '17

20 Nov '17

Hello, https://www.kamailio.org/wiki/devel/config-engines has links to https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… for example configs. All the "kemi" stuff is gone. 404 Error. Where i found actuall (kemi / kamailio 5.0.x) samples? -- Kind Regards *Karsten Horsmann*

3 5

no event received for call disconnection
by Aleksandar Sosic 20 Nov '17

20 Nov '17

Hi Everyone, we're trying to implement cgrates in our kamailio nodes. Everything is working fine except for the fact that kamailio is not signaling the call end to cgrates for prepaid users... The versions, logs and confs are: ``` # 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 with gcc 5.3.1 ``` ``` # cgr-engine --version CGRateS 0.9.1~rc8 git+15a0793 (2017-10-23T12:15:40+02:00) ``` The kamailio conf regarding our problem is: ``` # Inform CGRateS about CALL_END (stop debit loops, perform accounting if desired in this way) route[CGR_CALL_END] { if $sht(cgrconn=>cgr) == $null { xlog("Charging controller unreachable"); exit; } $var(callDur) = $TS - $dlg(start_ts); evapi_relay("{\"event\":\"CGR_CALL_END\", \"callid\":\"$dlg(callid)\", \"from_tag\":\"$dlg(from_tag)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_answertime\":\"$dlg(start_ts)\", \"cgr_duration\":\"$var(callDur)\", \"cgr_supplier\":\"$dlg_var(cgrSupplier)\", \"cgr_disconnectcause\":\"$T_reply_code\"}"); } ``` The kamailio logs: ``` 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} ``` Thanks for your help, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

2 13

kamailio 4.4 - jansson_get_field parse error actions
by Jonathan Hunter 17 Nov '17

17 Nov '17

Hi Guys, I am looking to send calls to a default route should we have issues when connecting to our API which provides routing instructions. On scenario is handling an invalid JSON response , which would result in a parse error. Is it possible to capture when there is a failed to parse error when dealing with JSON responses to an http query in kamailio? In particular when using the jansson_get_field, I am looking to default to a route should their be a parse error, so I am forcing an erro, I get this in the logs, but can I grab to act on it? Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' On receipt of this I would then like to hit a default route. Many thanks Jon

1 1

SUBSCRIBE NAT Keep-Alive
by Asgaroth 17 Nov '17

17 Nov '17

Hi All, I'd like to ask the guru's out there their opinion on how to solve the nat problem for presence subscriptions. We have an issue with subscriptions that get timed out and the notify clears the subscription out of the active_watchers table due to the notify timing out, I think this is expected and the right way to do it. However, how do others out there ensure that their subscription nat states are kept alive? For registrations we have nathelper, but, what is the sanest way of doing the same for subscriptions? I was originally thinking of sending notifications through our registrar/lookup server, but, I'm not sure if that is the correct way to do it. Any thoughts, tips, tricks would be much appreciated. Cheers

3 3

Kamailio questions
by Wilkins, Steve 17 Nov '17

17 Nov '17

Hello, I am new to Kamailio and to the Kamailio Mailing list and hope I am using it appropriately. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. For the sake of the argument, my server name is myServer, and it's IP Address is 172.1.1.13; Asterisk is on a different server. Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 myServer /usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.1.1.13:59794 failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work; I can make calls and use sip tools which pass requests through the Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what at such a high port. Question: The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Note: Even with no rtpproxy, I get the same error mentioned above in the Issue section. Thank you,

2 4

dr_load_routing_info(): failed to add rule id 4 -> skipping
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i experiment with the drouting module with mysql-db. I try to route with "alphanumeric" stuff - like 'tetris' in sip://tetris@mytest.example https://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp31352844 The DB-Field is an 64 string like route-id. In dr_rules.route-id i can use "named-routes" if i leave the prefix empty or i fill in only numbers. With strings i got this error: ERROR: drouting [dr_load.c:555]: dr_load_routing_info(): failed to add rule id 4 -> skipping SQL Result for rule-id 4: # dr_rules # ruleid, groupid, prefix, timerec, priority, routeid, gwlist, description '4', '9000', 'tetris', '', '0', 'tetris', '4', 'testroute' Is drouting only for "number-routing"? -- Kind Regards *Karsten Horsmann*

3 6

best practice to dynamic route block based on $rd - Domain in R-URI
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i am now at the point that one kamailio is not enough :). So i decide to set another kamailio internal as sip-router based on drouting. My config is straight forward the example config with additional drouting stuff. Whats the best practice to route based on $rd with kamailio? I dont want to end with an bunch of if then else. Should be come from database. Pseudo Code: if ($rd == old.sbc.domain) { route[oldstuff]; } else { route[newstuff]; } Is there an module that i can use? -- Kind Regards *Karsten Horsmann*

1 1

How to use $var as arguments to module functions
by Duarte Rocha 16 Nov '17

16 Nov '17

Greetings, Is there a way to use the NATHELPER functions with a variable insted of a string with the value ? For example, I can do "fix_nated_sdp("11");" but Kamailio throws an error if i use "fix_nated_sdp("$(var(nat_fix)");" or any other variations. How should i proceed to be able to do this ? Thanks

2 1

play music while processing a http request
by 赵国杰 16 Nov '17

16 Nov '17

Hello guys, I have the following senario: ua1 calls ua2. when the INVITE reaches the kamailio, before foward to ua2, kamailio has to do a http request using “http_async_query”. However, the http request can take a really long time. I want the kamailio start playing music when the http request is sent and stop the music when the http response is received. My config file is like this: request_route { ... t_newtran(); http_async_query("url here", "HTTP_REPLY"); } route[HTTP_REPLY] { ... # forward the invite to ua2 } how do i complete the config? Thanks

2 3

Dmq_usrloc
by sofos koko 14 Nov '17

14 Nov '17

Hi ...i would like to connect two kamailio in diffwrent network so i can get call from one to another is any kamailio.cfg example to help me do it!

1 0

topos and topos_redis : error log -> ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store
by Abdoul Osséni 14 Nov '17

14 Nov '17

hello, After activated redis backend for topos module, I have some error logs. Can you help me to understand these logs? # kamailio -V version: kamailio 5.1.0-rc0 (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 14:18:08 Nov 14 2017 with gcc 4.9.2 Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Abdoul OSSENI

1 0

AVPOPS: is_avp_set/avp_check "name" parameter as variable.
by Asgaroth 14 Nov '17

14 Nov '17

Hi All, I was having a read of the AVP module documentation and I see mentioned that in future releases of kamailio that the AVP module may become deprecated in favour of using AVPOPS module. To this end, I started looking to switch some of AVP functions over to AVPOPS functions when I cam across an issue where I cannot have the name parameter of the avp to check a variable. I have some attributes defined for a domain which I load up using lookup_domain, one of these attributes is a dispatcher setid based on the request method being processed. So, with AVP I can do something like the following to see if it is set: if ( attr_exists( $_s(dm_ds_setid_$rm) ) ) The above works as expected, but if I try using something equivelent using avpops functions, something like: if ( is_avp_set("$avp(s:$_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(s:dm_ds_setid_$rm)") ) or if ( is_avp_set("$avp($_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(dm_ds_setid_$rm)") ) Then it fails the check even though the avp is set correctly, there are no errors in the logs so it all appears to be okay. Likewise, when checking with the following AVP function, it works: if ( attr_equals("$_s(dm_ds_setid_$rm)", "100") ) But, if I try with AVPOPS function, it fails (I'm guessing its bacause the avp cannot be found) if ( avp_check("$avp(s:$_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(s:dm_ds_setid_$rm)", "100") ) or if ( avp_check("$avp($_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(dm_ds_setid_$rm", "100") ) Are the AVPOPS functions able to take nested variables like the above? Is this a bug or am I using them incorrectly? I'm currently trying this out on kamailio v5.0.4. Any thoughts on the issue would be appreciated. Thanks

2 2

max_contacts does not work as expected
by Gholamreza Sabery 14 Nov '17

14 Nov '17

I have a Kamailio server that has: modparam("registrar", "max_contacts", 2) But recently I noticed some users have more than two records in the location table. For example one of the users has three different records with nearly the same information. The records are as follows. Is this behavior normal? (I am using Kamailio 4XX). *************************** 1. row *************************** id: 171296745 ruid: uloc-2c69-59db6c73-7ca-132d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: XpCnmaAhRF cseq: 43 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 14 *************************** 2. row *************************** id: 171296746 ruid: uloc-2c69-59db6c73-7f2-eb5d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: BzMKuyJN8C cseq: 50 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 23 *************************** 3. row *************************** id: 171296747 ruid: uloc-2c69-59db6c73-80f-541d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: 3p9P4PgZSv cseq: 51 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 15

1 0

Load Gateway according hostname
by Ashutosh chaubey 13 Nov '17

13 Nov '17

Hi, I am trying to find a way to load a specific gateway but instead of lcr_id i want it load gateways according to the hostname. I am passing Hostname as a header to kamailio so its easy to access that but unabale to load gateways according to it. Any help? Regards, Ashutosh Chaubey

1 0

using bcrypt passwd hashing
by Walter Martín Villalba 13 Nov '17

13 Nov '17

Hello, I did some searches online and talked to some colleagues and it seems Kamailio only supports the traditional HTTP digest authentication, which uses MD5. I would like to know if any of you has been successful in using bcrypt/scrypt/pbkdf2 passwd hashing, instead of MD5, which has been deemed as obsolete and insecure a long time ago. Perhaps you've written your own auth module, or just modified the config script to call some other credential checking routine using a custom python/perl script (I'm thinking of doing the latter, of nothing better is available). If any of you have done something like this, using bcrypt or any other current and secure hashing algorithm, I would appreciate some guidance. If you haven't, aren't you concerned about storing MD5 password hashes in your database? Note: if I can't find a good answer using this list, I will try the developer's list next. Thanks in advance, Martín.

4 5

Kamailio issue
by bazou2 bazou2 13 Nov '17

13 Nov '17

Hello, I have a kamailio version 4.4 with an Asterisk 13 ... and i need to make a call from a sip phone app.(i have a user subscribed on the kamailio to be used on the sip phone app). The weird thing that first time, the call works (I can see the sip messages working fine between the app kamailio and asterisk) but when i try again it doesn't work. SIP messages show that the kamailio and Asterisk keep send trying request between them and the loop of these kind of messages doesn't end.. I need to know if i am missing something in the configuration file of the kamailio or what? Regards Sent from Outlook<http://aka.ms/weboutlook>

2 2

Development open in master branch (to be v5.2.x)
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, git branch 5.1 was just created (to host the release series v5.1.x), therefore new features can now be pushed again in master branch. They will be part of the next future release, likely to be numbered 5.2.x. Any fixes that affect existing code in branches 5.1 or older version have to be backported - push first to master and then cherry pick -- see the contributing guidelines at: - https://www.kamailio.org/wiki/devel/git-commit-guidelines#backporting_commi… Many thanks to all contributors so far! Testing of branch 5.1 and giving feedback for it is very appreciated! 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

Branch 5.1 created
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, the GIT branch 5.1 has just been created, it will host the release series 5.1.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 5.1 origin/5.1 Hopefully in two-three weeks time frame the full release of 5.1.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 5.1. No new features can get in branch 5.1. Enhancements to documentation or helping tools, as well as kemi exports are still allowed. If you are not sure about doing or not a backport, ask on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

GeoIP2 module deb package
by Joel Serrano 12 Nov '17

12 Nov '17

Hi, I can't seem to find the geoip2 module for debian stretch (installing from packages, kamailio 5.0 branch). Is it not available in the repository? Thanks, Joel.

2 4

Kamilio for IMS and VoLTE Setup Questions
by Jack Folgolburg 11 Nov '17

11 Nov '17

Hello, I saw the Kamailio World 2017 presentation "Kamilio for IMS and VoLTE" (https://www.youtube.com/watch?v=Wdhcdv5JVPw). I would like to install the servers for evaluation and testing. The slides say to use Ubuntu 16.04, but there are no xenial packages at the ng-voice repository. Is it okay to install on Ubuntu 14.04, since there are trusty packages? I have done this and got through all the configuration prompts. After installation, is there any way to confirm that the site is operational? How to bring up the management console (i.e. what is local URL)? In the presentation, it looks like a commercial site at ng-voice is accessed. I understand that there is a REST API. Is this API documented anywhere? Maybe, the most important question is, after installation, what is the fastest way to add two users and test access from IMS clients? Thank you, James

1 0

183 acc records even if early_media equals to 0
by Marco Capetta 09 Nov '17

09 Nov '17

Dear All, I'm facing a strange problem with the call accounting module: even if in my configuration I have the parameter: modparam("acc", "early_media", 0) I can find some ACC records with sip_code 180 or 183. I investigated those cases and this issue seems to happen when an endpoint sends an 183 message immediately followed by a 200 OK. The issue seems started from kamailio version 4.4.1, but I cannot be really sure of this. Have you ever had a similar problem? Do you have any suggestions on how to solve it? Thanks Marco

2 11

t_relay_to_udp - 500 server error occurred (6/SL)
by Donat Zenichev 09 Nov '17

09 Nov '17

Hi all. I have a problem with t_relay_to_udp function. My topology: UAC -> tcp -> kamailio <-> udp <-> asterisk Kamailio processes requests from clients over TCP and relays requests to asterisk over UDP. Kamailio gets INVITE from client, relays it to asterisk, then he sends 100 trying to user client and abruptly sends 500 internal error to client (after 100 trying). But the main interesting thing is, that asterisk answers on INVITE and dialog continues between kamailio and asterisk (without user agent client) - it reaches the destination then, without originator : ) How error on kamailio looks like: Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: tm [t_fwd.c:1716]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (6/SL) The same problem is, when I change transport protocol between kamailio and asterisk to TCP, using the t_relay_to_tcp function. When I use t_relay function, it works fine, nothing happens that can drop dialogs between kamailio and clients. Relaying code part: if ($var(to_asterisk)=="1") { xlog("L_NOTICE","Relaying to asterisk - change socket to udp:10.1.1.1:5070 \n"); force_send_socket(udp:10.1.1.1:5070); if (!t_relay_to_udp()) { sl_reply_error(); } } I need to use t_relay_to_udp with force_send_socket, because kamailio changes the source port (source socket) every time he generate new INVITEs to asterisk and this is irrelevant. Dump between kamailio and user agent client: INVITE sip:1111111111example@kam1.domain.com:5070;transport=TCP SIP/2.0 Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z- Max-Forwards: 70 Contact: <sip:debug.device-example@uac.domain.example:5065;transport=TCP> To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="debug.device-example",realm=" kam1.domain.com ",nonce="WgQGlVoEBWnhH3CF6DE77uKxKJx0tT4e",uri="sip:1111111111example@kam1.domain.com:5070 ;transport=TCP",response="4753aa849715d26d0dde1b3c2314c843",a orithm=MD5 Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 v=0 o=Z 0 0 IN IP4 uac.domain.example s=Z c=IN IP4 uac.domain.example t=0 0 m=audio 8000 RTP/AVP 8 3 110 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: MS Lync Content-Length: 0 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL) Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070 ;transport=TCP>;tag=9a78ee8ed9a3d4ba441c41d83ba2e904.4113 From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: example Content-Length: 0 Thanks in advance for hints. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-800 http://wnet.ua

1 0

Topoh module configuration for SIP proxy and SIP registrar
by Christian Conus 08 Nov '17

08 Nov '17

Hello, I am trying to configure the topoh module for kamailio 5.0.2. I read the documentation but some points are still not clear to me. The topology of my network is as follow: - Two SIP proxies - Two SIP registrars - Only one SIP proxy is running (the other take over automatically in case the first fails) - Both registrars are running (proxy sends requests to both registrars using round robin algorithm) Text mode graph of the topology (hope it helps understand): I tried to enable topoh on SIP proxy only, on both proxy and registrars, with same mask_key everywhere and with different mask_key between proxy and registrars but everytime I could still see parts of the SIP header which was not obfuscated. My questions are: - Is it necessary to enable topoh module on each SIP proxy and SIP registrar ? - The mask_key should be the same on each server or it should be different between proxy and registrars? - the mask address should be the same on each server or it should be different between proxy and registrars ? Thanks in advance for your help, Christian -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 10

evapi_async_relay() failed
by Abdul Basit 07 Nov '17

07 Nov '17

Hi list, I am configuring CGRateS with kamailio 4.4 using deb packages on debian 8. I am facing a issue while processing event_route[dialog:start] for answering call INVITE at evapi_async_relay("{\"event\":\"CGR_CALL_START\", with following error /usr/sbin/kamailio[13887]: ERROR: evapi [evapi_mod.c:288]: w_evapi_async_relay(): failed to suspend request processing However, after discussion with Dan on IRC i replaced function evapi_async_relay() with evapi_relay() resolved the issue. Whats causing this error? Can anyone suggest some way out for keep using evapi_async_relay? -- regards, abdul basit

2 1

Kamailio SIP-I
by Saleem Manasfi 07 Nov '17

07 Nov '17

Dears, Please if you can let us know how does Kamailio reads the SIP-I body (ISUP)? Is there any document or link that may help?

2 1

Re: [SR-Users] Regarding Opensource DID Routing Solution using Kamailio
by surendra tiwari 06 Nov '17

06 Nov '17

Hi All, I have used the kamailio and redis to create the DID routing solution.please find below in github.this repository is created to help the person who want to setup there own DID routing solution with following feature: 1. Inbound Termination with Carrier IP validation. 2. Carrier LCR for DID/TFN to PSTN forwarding. 3. Inbound Abuse Block. 4. CDR in MongoDB. 5. IPTables Block for Sip-Scanners. 6. Integration with RtpEngine. 7. RedisDB for quick DB Access. https://github.com/surendratiwari3/KamInboundSIP On Mon, Nov 6, 2017 at 9:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > your email was addressed to kamailio-announce mailing list (I took your > message from its bounces filter), but that list is only for announcement > issued by Kamailio project (major news about our releases, security > incidents, events, etc...). > > News from community members are welcome and can be sent to: > > - sr-users(a)lists.kamailio.org for non-commercial announcements > > - business(a)lists.kamailio.org for announcements related to commercial > offerings/products (but here you can also send non-commercial announcements) > > Your announcement is not commercial and can be sent to sr-users. Once it > will be there, I will also make a news article on kamailio.org website. > > I didn't have the time yet to check in detail what you put together, but > sounds interesting. > > Thanks for sharing, > Daniel > > (cc-ed Fred as he is moderator for kamailio announce mailing list) > > On 01.11.17 18:46, surendra tiwari wrote: > > Hi All, > > I have used the kamailio and redis to create the DID routing > solution.please find below in github.this repository is created to help the > person who want to setup there own DID routing solution with following > feature: > > > 1. Inbound Termination with Carrier IP validation. > 2. Carrier LCR for DID/TFN to PSTN forwarding. > 3. Inbound Abuse Block. > 4. CDR in MongoDB. > 5. IPTables Block for Sip-Scanners. > 6. Integration with RtpEngine. > 7. RedisDB for quick DB Access. > > > https://github.com/surendratiwari3/KamInboundSIP > > -- > With Regards, > Surendra Tiwari > VoIP Application Developer, > +91-9967609476 <+91%2099676%2009476> > > > -- > Daniel-Constantin Mierlawww.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 > > -- With Regards, Surendra Tiwari VoIP Application Developer, Plivo Comm +91-9967609476

1 0

Sipcapture HEP Values
by Soltanici Ilie 06 Nov '17

06 Nov '17

Hi all, How can i get the Destination IP from HEP packet? I tried $hep(dst _ip) like in documentation (https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#sipcapture_mo…) - but is not working. (pvar "hep" has an invalid name param [dst_ip]). Thank You.

1 0

Invalid JSON when dumping htable with JSONRPC-S
by Marrold 03 Nov '17

03 Nov '17

Hi All, I am currently trying to retrieve the contents of a htable using the JSONRPC-S module over HTTP, however I have observed if a slot contains multiple values, the JSON returned has duplicate keys which could be considered invalid. Here are the relevant lines from kamailio.cfg - event_route[xhttp:request] { if ($hu =~ "^/carrier") { jsonrpc_exec('{"jsonrpc": "2.0", "method": "htable.dump", "params" : {"htable": "carrier"}, "id": 1}'); xhttp_reply("200", "OK", "text/html", "$jsonrpl(body)"); } } And the output, showing a slot with two items- { "entry": 5, "size": 2, "slot": { "item": { "name": "carrierA", "value": "90", "type": "str" }, "item": { "name": "carrierB", "value": "190", "type": "str" } } } And the same from `kamcmd -s unix:/tmp/kamailio_ctl htable.dump carrier` - { entry: 5 size: 2 slot: { item: { name: carrierA value: 90 type: str } item: { name: carrierB value: 190 type: str } } } Would it be possible to change it to something like the following - { "entry": 5, "size": 2, "slot": [ { "name": "carrierA", "value": "90", "type": "str" }, { "name": "carrierB", "value": "190", "type": "str" } ] } Any other suggestions or work arounds would be appreciated, Thanks Matthew

2 1

Contacts in 302
by Diego Nadares 03 Nov '17

03 Nov '17

Hi Guys, I'm generating a 302 reply from kamailio. In this 302 I append new branches with new contacts. if($var(routing)=~"redirect"){ jansson_get("contacts_len", "$var(evmsg)", "$var(contacts_len)"); xlog("L_INFO", "Contacts len $var(contacts_len)"); $var(i) = 0; while ($var(i) < $var(contacts_len)){ jansson_get("contacts[$var(i)]", "$var(evmsg)", "$var(contact)"); append_branch($var(contact), "0.5"); $var(i) = $var(i) + 1; } send_reply("302", "Moved Temporarily"); exit; } The problem i'm facing is that I can't delete the original contact *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>* The resulting conctact: Contact: *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>*, <sip:1111@188.111.111.112>;q=0.5, <sip:1111@188.111.111.112>;q=0.5 I all ready tried with remove_hf('Contact') and adding the new ones after that but it doesn't worke either. The problem is that one gw takes the first contact over and over again and never the other two. Thanks in advance. Diego.

2 4

Multiple IPs
by Oz Mortimer 03 Nov '17

03 Nov '17

Hi all, I have Kamailio set as an SBC with freeswitch behind - that all works fine. Freeswitch can send calls out VIA Kamailio to external IP address’, but if I try sending a call to another IP hosted by Kamailio the seems to go nowhere… Supplier -> Kamailio -> Customer1 -> Kamailio -> Customer2 The Kamailio server has multiple assigned IPs, which are used to multi-tennant - but I cant for the like of me to get Tennant to tenant calls to work (via freeswitch) Any pointers?

2 13

onreply_route block not processing replies
by José Seabra 03 Nov '17

03 Nov '17

Hi there, I'm facing an issue regarding with replies coming to kamailio that aren't processes inside of onreply_route block. Anyone here can help me understand why these replies (1XX) aren't entering on onreply_route bock? is there any situation already identified on Kamailio that can originate this behavior? The Kamailio version used is 5.0.1. Thank you Regards -- José Seabra

2 3

Branching Kamailio v5.1
by Daniel-Constantin Mierla 03 Nov '17

03 Nov '17

Hello, I am planning to create the got branch 5.1 (to be used for v5.1.x series) by end of next week (likely on Friday, Nov 10, 2017). After that, the master will be open for new features and fixes will need to be backported to 5.1. Should anyone want a different schedule, reply to the mailing lists and propose your plans. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

WebRTC error code 488 in iOS application
by Denys Pozniak 03 Nov '17

03 Nov '17

Hello! Probably this case is not connected Kamailio directly, but it can be somebody will point me in the correct direction :) We have VoIP solution: FreeSWITCH -> Kamailio + Rtpengine -> WebRTC Chrome. Everything works Ok except iOS application based on React Native WebRTC ( https://github.com/oney/react-native-webrtc) + JsSIP. It answers error 488 on incoming INVITE (wSDP): INVITE SDP to iOS: *......* *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* iOS console log: *11:55:23.670 browser.js:133 JsSIP:WebSocketInterface send() +10ms* *11:55:23.670 browser.js:133 JsSIP:RTCSession session progress +5ms* *11:55:23.671 browser.js:133 JsSIP:RTCSession emit "progress" +1ms* *11:55:23.671 webrtc.js:108 WebRTC.on progress: local* *11:55:23.679 browser.js:133 JsSIP:RTCSession emit "peerconnection" +8ms* *11:55:23.680 webrtc.js:89 WebRTC.on peerconnection* *11:55:23.680 browser.js:133 JsSIP:RTCSession emit "sdp" +1ms* *11:55:23.689 browser.js:133 JsSIP:Transport send() +20ms**11:55:23.689 browser.js:133 JsSIP:Transport sending message:* *SIP/2.0 488 Not Acceptable Here* *11:55:23.701 debug.js:115 JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:Error: Failed to set remote offer sdp: Failed to create channels.* I found similar issue on thier git https://github.com/oney/react-native-webrtc/issues/293 and main solution is to struct SDP in way like: *a=group:BUNDLE audiom=audio PORT RTP/SAVPF N M Ka=mid:audio* How to modify SDP on Kamailio side (after Rtpengine modification) like below? *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *a=group:BUNDLE audio* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=mid:audio* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* BR, Denys

1 1

User Registration with location awareness
by Daisy Stevens 03 Nov '17

03 Nov '17

Hi Guys Please forgive me if I seem to asking stupid questions but I am a complete dummy at the moment regarding this software. I would love to be able to set up two kamailio servers with the idea of expansion in the future to handle SIP registrations, however having a two server model how would I get the calls to route to another user that is registered on the other kamailio box? Kind Regards Daisy -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

2 1

msilo Content-Type for picture and video?
by sscc 03 Nov '17

03 Nov '17

Hello kamailio folks i am successfully storing and dumping messages for off line users using msilo content type text/plain. i want to store off-line messages for pictures and videos. my particular queries are 1) is msilo store picture and videos? (i have observed blob in silo database table which is used for pictures. 2) in case it does what will be declared in content type hdr parameter? any help in this regard will highly be appreciated. regards, sscc -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Re: [SR-Users] Reading the P-Assserted-identity header
by Gertjan Wolzak 02 Nov '17

02 Nov '17

Alex and Daniel, Thank you for the feedback, the tip from Daniel helped me pinpoint the problem. And the the one from Alex showed that. The problem was between keyboard and chair, I should have seen that the SBC removed the PAI. Once I changed that, the PAI arrived at Kamailio and manipulation was a breeze. Thanks guys.

1 0

Tcp socket error with active/active kamailio schema
by Donat Zenichev 02 Nov '17

02 Nov '17

Hi guys. I'm trying to realize active/active schema with kamailio. Topology is as following: active/active kamailio -> routing server (asterisk) Kamailio servers work as registrar/location servers and sbc. Asterisk server performs routing functions for calls. We use sip / sips / ws / wss for signaling. Schema built on aws platform and both kamailio have virtual ip address. So when first kamailio crashes, second one assigns ip address of first kamailio. And vice-versa. Both kamailio listens on both shared addresses all the time (with help of sysctl non lolcal bind function). Problem occurs when first kamailio goes down and second kamailio gets ip address of first one. Asterisk tries to send in-dialog requests (for dialogs that were established on first kamailio), request goes to second kamailio and asterisk gets a socket error. And this is proper behaviour I think, because first kamailio established tcp connection with asterisk from certain port, that is not listened on second kamailio. So that, I think that the solution is to use force_send_socket function, but I'm not confident about it. If anyone has ideas, please send your advices. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Kamailio not calling each time
by hanoulia salameh 02 Nov '17

02 Nov '17

Hello, I have a problem with kamailio (v4.4), when i try to call from a sip phone through kamailio, the call is interrupted and hang up automatically but after 4s it continues to retry (I can see in the log file Session Progress) and goes through the asterisk (certified/13.13-cert4) and establishes a normal call (It rings at the client side) while it's still disconnected from the sip phone side But when i try again it works fine sometimes. Am I missing something in my configuration file? What can I show in order to help me? Regards

2 2

Kamailio Realtime Integration with Asterisk
by Floimair Florian 02 Nov '17

02 Nov '17

Hi! I found a few tutorials from Daniel on this topic , the latest being https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astd… . Has anyone ever tried this with a more recent Asterisk version (13 or newer) and using chan_pjsip instead of chan_sip? I'd be willing to update the tutorial once I get it running but I would like to avoid going through obstacles that were already handled ;-) My desired setup is 2 Asterisk boxes (which share configuration, useres and registrations via DB) behind a Kamailio as Load-balancer (got that working already). With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 f.floimair(a)commend.com<mailto:f.floimair@commend.com> http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg

1 0

Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 02 Nov '17

02 Nov '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

4 11

Reading the P-Assserted-identity header
by Gertjan Wolzak 01 Nov '17

01 Nov '17

Hello Kamailions, I receive anonymous calls where the PAI has a prefix which I need to strip before forwarding the call to a proxy. I am trying to read the the header with $hdr(P-Asserted-Identity) and I tried the $ai pseudovariable, but I get an empty reply back every time. xlog("PAI is $ai"); results in ; "Nov 1 16:19:55 proxy04 /usr/local/sbin/kamailio[14814]: ERROR: <script>: PAI is <null>" Same for xlog("PAI is $hdr(P-Asserted-Identity). This works for the Privacy header but not the P-Asserted-Identity header. Hope someone can tell me what I am doing wrong.... Rgds, Gertjan

3 2

Record-Route IP Value
by Soltanici Ilie 31 Oct '17

31 Oct '17

Hi All, Having the following Header: Record-Route: <sip:192.168.1.1;lr;did=637.07c7c2d7> How can i extract only IP value from this header? Thank You.

2 3

module version mismatch for /usr/local/lib64/kamailio/modules/kex.so
by Abdul Basit 31 Oct '17

31 Oct '17

Hi, My kamailio installation with 5.0.3 was working fine. I update kamailio source to 5.0.4 and make all; make install I started receiving error below: /etc/init.d/kamailio start [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(7426) ERROR: <core> [core/sr_module.c:431]: version_control(): module version mismatch for /usr/local/lib64/kamailio/modules/kex.so; core: kamailio 5.0.3 (x86_64/lin[FAILmodule: kamailio 5.0.4 (x86_64/linux) ... failed! failed! kamailio.cfg is same as earlier. I removed kamailio sources and modules folder /usr/local/lib64/kamailio/modules/ I also dropped old kamailio postgres database and created new one after fresh source install of version 5.0.4 from git. # git clone https://github.com/kamailio/kamailio kamailio # cd kamailio # git checkout -b 5.0 origin/5.0 But this error persists. Whats the resolution? Is this a bug? -- regards, abdul basit

2 2

Source comparisons by DNS hostname
by Alex Balashov 31 Oct '17

31 Oct '17

Hi, Is there a reasonable way, in Kamailio, to determine whether the source of a given SIP message is equivalent to the IP address to which a given DNS hostname resolves? i.e. if(src_ip == 'fqdn.domain.xyz') but not src_ip, obviously. I can't find any core keywords that would seem to fit the bill for a composite comparison. Any other options? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Radius accounting and cdr's
by Jan Hazenberg 31 Oct '17

31 Oct '17

Hi all, I have a question regarding radius accounting. I would like to have a config where i store local cdr records (and not the ACC records) and send accounting to radius as well. I have the following config: # ----- acc params ----- /* General settings */ modparam("acc", "log_level", 0) modparam("acc", "log_facility", "LOG_LOCAL2") /* Enable cdr's */ modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_LOCAL2") modparam("acc", "cdr_on_failed", 1) /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) /* Accounting triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACCMISSED) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") This works but stores the raw accounting records as well. If i disable the acc log_flag it stops sending radius accounting records (while the acc_radius log_flag is still enabled). Is it possible to send only radius accounting records and disable the local logging of those records, but leave the cdr logging enabled? Or is there a option where i can set only the cdr_log_level so accounting records will only show up when i increase the debug level? Another option is to use a seperate log facility for accounting and cdr's, but it would be nice if it is possible to disable the logging of accounting records and only keep the cdr's. I'm running kamailio 5.0.3. Thanks, Jan

2 2

Re: [SR-Users] Kamailio for Unified Communication
by Angel Elena 31 Oct '17

31 Oct '17

kamailio is a "proxy" server.... this features may be added by a media-server... freeswitch, asterisk, seems. You can put kamailio as "sip proxy" and you can add these services behind kamailio proxy -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Zarafa Webapp. -----Mensaje original----- De: Jayanth Acharya <jayachar88(a)gmail.com> Enviado: Mar 31-10-2017 10:40 Asunto: [SR-Users] Kamailio for Unified Communication Para: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>; > Hi, > > Can Kamailio be used to provide following broad-type of services ? > > 1. Unified Communication services > 2. Contact Center services > 3. Instant Messaging / Chat services > 4. Voicemail service > 5. Presence aware communication services > 6. Desktop sharing service > > For many/some of the above, I'd expect back2back user-agent capabilities to be > mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are > there some modules that may be used for this purpose ? > > thanks, > Jay > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > >

1 0

Kamailio for Unified Communication
by Jayanth Acharya 31 Oct '17

31 Oct '17

Hi, Can Kamailio be used to provide following broad-type of services ? 1. Unified Communication services 2. Contact Center services 3. Instant Messaging / Chat services 4. Voicemail service 5. Presence aware communication services 6. Desktop sharing service For many/some of the above, I'd expect back2back user-agent capabilities to be mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are there some modules that may be used for this purpose ? thanks, Jay

1 0

Reinvite/ACK race
by Alex Balashov 30 Oct '17

30 Oct '17

Hi, I've got a scenario like so: UA A -----> Kamailio P ----> UA B 1. UA A initiates call through Kamailio P; 2. Dialog is established and confirmed, with Record-Route; 3. UA B sends reinvite #1 through P to A; 4. UA B sends 2xx reply; 5. UA B sends end-to-end ACK for reinvite #1 and almost simultaneously sends reinvite #2. The temporal delta is between reinvite #2 and ACK for reinvite #1 on the wire is 3 ms. So, the result — for all kinds of stochastic processing and userspace scheduling type reasons — is that the reinvite is forwarded first, before the ACK. That leads to a 500 / 491 scenario UA A. The cause, as we all know, is that Kamailio's worker threads are loosely coupled, and incoming UDP datagrams are distributed directly by the kernel. Is there any general guidance on what to do with these scenarios? I looked at RFC 5407 § 3.1.4, which appears to describe a similar, but not identical scenario involving an initial INVITE and subsequent reinvite. As far as I can tell, the recommendation in that standard is "space the messaging out more in time". Switching to TCP would presumably help, since any given flow would involve a single connection to a single worker thread and the transport would guarantee ordering. However, that's not really feasible in this implementation for a host of reasons. I know Kamailio has some config locking primitives, but I am extremely wary of complex synchronisation on dialog-identifying attributes, particularly if there is a possibility that such locks could stall a worker perpetually. Any other thoughts welcome! Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

unexpected behavior of save from registrar module
by Vasiliy Ganchev 30 Oct '17

30 Oct '17

Hi, community! I have in my Kamailio script following block (while REGISTER processing, after auth succeed): ..... /* Save to usrloc */ lock("LOCATION_LOCK"); if (!save("location")) { unlock("LOCATION_LOCK"); sl_reply_error(); exit(); } unlock("LOCATION_LOCK"); ...... According to module documentation - if "save" - failed (for any reason) - I expected the control to be returned to the script with error code "-1" But, there are cases - when this does not happen - which lead to deadlock for workers processing REGISTERs (I never unlock("LOCATION_LOCK")): - module answers with internal error - and return 0 from "save" function: https://github.com/kamailio/kamailio/blob/master/src/modules/registrar/save… - line 1013 My use case happened due to malfunctioning device that in some moment of time: - send REGISTER update - kamailio answers 200OK - device retransmit the same REGISTER (same CSeq) - during 20 seconds (hardcoded value in registrar module) - kamailio answers 200OK (and everything is ok) - if retransmission continue more than 20s - kamailio answered 400 Bad request, and did not return control to script (as a result I had no chance to unlock("LOCATION_LOCK") from code above) Tested it on kamailio 4.2.5 and 5.0.3. as for me, possible solutions are: - return -1 instead of 0 from "save" by default - if there are some reasons not to do this - we should change documentation, and explicitly mention that there are cases - when script execution is stopped. thanks in advance! cheers -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

3 5

sanity passes invalid uri
by Juha Heinanen 30 Oct '17

30 Oct '17

I noticed that K 5.0 sanity() test passes uri that contains ` (back quote) character. Config: modparam("sanity", "default_checks", 1024) /* URI checks */ modparam("sanity", "uri_checks", 3) /* RURI, From */ xlog("L_INFO", "Checking $ru\n"); if (!sanity_check()) xlog("L_INFO", "Check failed\n"); else xlog("L_INFO", "Check passed\n"); Syslog: Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Checking sip:jh@te`st.fi Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Check passed According to RFC3261: hostname = *( domainlabel "." ) toplabel [ "." ] domainlabel = alphanum / alphanum *( alphanum / "-" ) alphanum Is this a bug or am I missing something? -- Juha

2 2

Remove all cpecial characters (\n,\t etc)from xml body
by Yuriy Gorlichenko 30 Oct '17

30 Oct '17

Hi. I building external presentity service for web based phones Main trouble that i have at the xml body of PUBLISH message \n \t characters and because of it can't set body as string to the redis So I trying to find a way to remove special characters form the xml body for get oneline string instead of multiline I use native .cfg file for now and, unfortunattely, can not use lua or phyton I tried $var(body)=$(rb{s.rm,$var(RemoveThisStaff)}) where $var(RemoveThisStaff) = "\n" and "\t" but it was unsuccesfull. Maybe someone will help to kno how to do it.

2 2

Issue with sipcapture
by Nathan Leyton 28 Oct '17

28 Oct '17

Hi There I am new to Kamailio and have just installed and got it up and running as a sipcapture node. It is working well and logging all the hep sip traffic in my homer database. I have an issue with rtcp extracted to the msg column is not json. It is some kind of weird encoded string (not base64), this is not readable by homer in any way. When I use nonsip_hook I am able to extract the json from the hep and put it in the msg field. However when I return 1 from the hep route on sip to go back to main route I get parsing errors like so on all the sip messages:- Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: sipcapture [sipcapture.c:2711]: nosip_hep_msg(): couldn't parse sip message Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/receive.c:186]: receive_msg(): core parsing of SIP message failed (178.78.120.2:44389/1) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [(null)] Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: WARNING: <core> [core/receive.c:193]: receive_msg(): parsing relevant headers failed Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/msg_parser.c:675]: parse_msg(): ERROR: parse_msg: message=<HEP3#002,> It never actually enters the main route. With nonsip_hook off, I enter the main route and all my sip messages are there in homer, just missing rtcp as the data is a weird encoded string. Hope someone can help Thanks Nathan -- Nathan Leyton / Chief Technology Officer nathan(a)callhandling.co.uk / 07482 917 412 Call Handling Ltd Office: 0333 321 0888 / Fax: 0333 321 0877 Bakerloo Chambers304 Edgware Road, London, W2 1DY www.callhandling.co.uk [image: htmlsig.com] <https://htmlsig.com/t/0000001BREBFR>

2 1

fix_nated_contact and IPv6
by Sebastian Damm 27 Oct '17

27 Oct '17

Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: <sip:bob@192.168.8.132> outbound Contact: <sip:bob@1234:1234:0:1234:0:0:0:2> The correct way of transforming the Contact URI would be: Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]> This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian

4 4

Too Long Contact URI
by Soltanici Ilie 27 Oct '17

27 Oct '17

Hi All, I cannot Register some softphones because of too long Contact header when using TCP protocol. Using UDP - everything works ok and this is how the Contact header looking: sip:XXXXXXXXX@XXX.XXX.XXX.XXX:64773;rinstance=6502234ef65ea6e7 But, switching to the TCP i'm receiving the following warning: WARNING: registrar [sip_msg.c:194]: check_contacts(): contact uri is too long:[sip:XXXXXXXXX@192.168.0.1:60471;rinstance=bf4d7aae0f099bf0;transport=tcp] How can i solve this issue? Thank You.

1 1

How to send OPTIONS message by kamailio server
by 吴妙灵 27 Oct '17

27 Oct '17

Hello, I don't know how to send OPTIONS message from kamailio server to user client,can you give me same advice? Cheers, Ada

4 3

Timer function didnot trigger the routing function
by RAJA SREE RATHNA gomathinayagam 27 Oct '17

27 Oct '17

Hi, I tried using the timer function to call a function after specific time but it was not working as specified. Could you please help me with some example using timer to trigger a function. I tried it using the description given in below link. www.kamailio.org/docs/modules/devel/modules/timer.html Thanks and regards, Rathna.

2 3

Kamailio v5.0.4 Released
by Daniel-Constantin Mierla 27 Oct '17

27 Oct '17

Hello, Kamailio SIP Server v5.0.4 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.0.x versions are strongly recommended to be upgraded to v5.0.4. For more details about version 5.0.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/10/kamailio-v5-0-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 4

Set timeout before 180/183 reply arrived
by Yu Boot 27 Oct '17

27 Oct '17

Hello. Have a problem with some uplinks. They sometimes respond "100 Trying" on initial INVITE and... that's all. A caller waits for about 30 seconds and cancel a call deciding there's something wrong with our network. How to use failure route when 180/183 answers don't arrive in 5 or 10 seconds after INVITE?

2 2

Get value of a custom header param
by Arsen 26 Oct '17

26 Oct '17

Hi guys! Is it possible to get value of a custom parameter of the authorization header? I've checked selects in the textops mod and try the following: if (is_present_hf("Authorization")) { xplog("L_ERR", "custom auth $sel((a)hf_value.authorization.edatech_opaque) \n"); } Arsen Semionov www.eurolan.info cell: +442035198881

3 5

Upgrading from 1.5 to 3.x
by Brandon Shiers 26 Oct '17

26 Oct '17

I have a client that has an old Kamailio 1.5 installation. They would like to migrate and get off their old hardware and old OS. I've read the upgrade page for the 1.5.x migration to 3.0.0, are there any gotchas to be aware of other than what's there? Brandon Shiers, Network Engineer 937 West Main Street Riverton, WY 82501 307.857.6704 (o) 307.840.2366 (c) 307.856.1499 (f) brandon.shiers(a)cerento.com<mailto:Brandon.Shiers@cerento.com> [http://www.cerento.com/emailsignatures/logo.jpg]

2 1

Regarding rtimer usage under specific condition
by RAJA SREE RATHNA gomathinayagam 26 Oct '17

26 Oct '17

Hello, While going through the rtimer module we noticed that it execute the routing function repeatedly after the given interval of time. But our specification is to execute the routing function only once and timer should be enabled based on specific condition. Also there is no enable or disable option for rtimer module as in timer module, whether rtimer module can be enabled based on specific condition.If it is possible could you please suggest the solution. Thanks & Regards, Rathna.

2 1

shared dialogs
by : Paolo Visintin - Time-Net S.r.l. 26 Oct '17

26 Oct '17

Hi kamailio users, I'm wondering if there's something already done with shared dialog among more kamailio instances, in order to manage branches initalized by another kamailio instance. A ctually seems not possible (also with db mode) because kamailio is looking into caller/callee_sock and if it's different does not manage in a "standard" failover environment (master / slave) with vIP and keepalived and $fs = VIRTUAL_IP everything is working fine but in a distributed environment this could not happen as the relay is managed by a kamailio proxy Thanks in advance Paolo

4 7

I-CSCF Prefixes IMPU with + Symbol
by Ron McLeod 25 Oct '17

25 Oct '17

I am working with v5.0.3 and the IMS extensions. When I attempt to send a SIP MESSAGE from the application server towards the UE, the I-CSCF used to determine the S-CSCF for the subscriber by sending a Location-Info request to the HSS. I found that this request was failing because the IMPU (IMS Public User Identity) was being mangled by the I-CSCF, prefixing the user part with a + symbol. The example configuration file for the I-CSCF does in-fact does cause the r-uri to be prefixed with a + symbol if one is not already present. I'm wondering why since it breaks the LIR/LIA lookup. AS I-CSCF HSS | MESSAGE | | | ---------------------> | | | sip:ron@provider1.test | | | | | | | | | | LIR | | | ----------------------> | | | sip:+ron@provider1.test | | | | | | | | | DIAMETER_UNKNOWN_USER | | | <---------------------- | | | | | | | | 403 | | | <--------------------- | | | FORBIDDEN | | route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } ... Thanks, Ron

1 0

Testing for user=phone with siputils uri_param() doesn't work
by George Diamantopoulos 25 Oct '17

25 Oct '17

Hello, In my script I've had to test for the user=phone parameter in URIs. I thought uri_param() from siputils would be handy for this operation on the RURI, so I did: if ( !uri_param("user","phone") ) { add_uri_param("user=phone"); } Unfortunately this wouldn't work (uri_param returns -1), and I end up with double user=phone params in the RURI. I was able to work around this by changing the test to: if ( !($(ru{uri.uparam}) == "phone") ) { add_uri_param("user=phone"); } But I wanted to ask if this is intentional or if I should file a bug or something. Thanks. George

3 3

Planning Kamailio v5.0.4
by Daniel-Constantin Mierla 25 Oct '17

25 Oct '17

Hello, I am considering to release Kamailio v5.0.4 later this week, likely on Wednesday or Tuesday. If you are aware of any issue in branch 5.0 which is not yet reported on github.com/kamailio/kamailio issues tracker, then do it as soon as possible to have a chance to look at it. 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

3 5

usrloc module does not work with db_cluster
by Uri Shacked 25 Oct '17

25 Oct '17

Hi, I started using db_cluster module. DB are master-slave replication both ways. #!define DBURL "cluster://kamailio" when i stop the primary mysql server, all subscribers fails to register again. seems like the usrloc module does not work with the cluster. modparam("usrloc", "db_mode", 1) modparam("db_cluster", "cluster", "kamailio=>primary=9s9s;secondary=8s8s") it works OK for modules like dispatcher and htable to reload from DB. it shows the sql driver errors, but reloads from the secondary server. I need auto expire, so, p_usrloc is not a good option. working with version 4.2 any ideas ? Uri

2 1

FW: Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 25 Oct '17

25 Oct '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

1 0

Advice on taking the best path to build a Presence & IM server
by José Seabra 24 Oct '17

24 Oct '17

Hello there, I'm writing here in order to get your experiences about a Presence & IM server. I would like to know what is the best solution to follow: - SIP Simple or XMPP? I've more knowledge about SIP Simple with kamailio but i did some researches and i found several open sources unified communications software that's implement Presence and IM using an XMPP server. So what is your opinion about which is the best solution to choose, taking in considerations the following aspects: 1. scalability 2. interopeability 3. permornace 4. features(like: Personal assistance, voicemail integration, chat group, integration with social media) Should i choose Kamailio SIP Simple with xcap or an XMPP server(Prosody or ejabberd, etc) I will appreciate a lot your opinion. Thank you Regards José -- Cumprimentos José Seabra

3 4

lcr
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

(no subject)
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

How to check 4XX/5XX response in general.
by vivek＠advaitamtech.com 24 Oct '17

24 Oct '17

Hi, Is there any way to catch all 4XX and 5XX SIP error codes rather then specifying individually ie. 404, 500, 400 etc in kamailio. Thanks, Vivek.

2 2

NatHelper ignoring force_socket module parameter
by Asgaroth 24 Oct '17

24 Oct '17

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, 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: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

2 5

No music on hold with mohqueue
by Fedor 24 Oct '17

24 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 rttproxy configured 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

Multi crashes in the same day
by igor.potjevlesch＠gmail.com 23 Oct '17

23 Oct '17

Hello ! I have multi crashes on a Kamailio 4.2.8 instance. Already 6 coredump today, same time with the same scenario. Below the output of the last coredump: (gdb) where #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 #2 0x000000000061aad2 in qm_debug_frag (qm=0x7f9a8a7aa010, f=0x7f9a8aca5968) at mem/q_malloc.c:161 #3 0x000000000061bf1c in qm_malloc (qm=0x7f9a8a7aa010, size=24, file=0x7131d0 "<core>: re.c", func=0x71483e "subst_run", line=440) at mem/q_malloc.c:388 #4 0x0000000000505e2c in subst_run (se=0x7f9a8ac08b78, input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, count=0x0) at re.c:440 #5 0x0000000000506fff in subst_str (input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, se=0x7f9a8ac08b78, count=0x0) at re.c:515 #6 0x00007f9a87ff629d in subst_uri_f (msg=0x7f9a8abfef38, subst=0x7f9a8ac08b78 "X\215\300\212\232\177", ignored=0x0) at textops.c:744 #7 0x000000000041d51b in do_action (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1094 #8 0x0000000000429ba3 in run_actions (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1583 #9 0x000000000042a208 in run_actions_safe (h=0x7fffed081380, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1648 #10 0x0000000000542994 in rval_get_int (h=0x7fffed081380, msg=0x7f9a8abfef38, i=0x7fffed080890, rv=0x7f9a8ab64ef8, cache=0x0) at rvalue.c:924 #11 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed081380, msg=0x7f9a8abfef38, res=0x7fffed080890, rve=0x7f9a8ab64ef0) at rvalue.c:1918 #12 0x000000000041cf77 in do_action (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1064 #13 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1583 #14 0x000000000041d3f6 in do_action (h=0x7fffed081380, a=0x7f9a8ab6b070, msg=0x7f9a8abfef38) at action.c:1079 #15 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab607b8, msg=0x7f9a8abfef38) at action.c:1583 #16 0x000000000042a2d0 in run_top_route (a=0x7f9a8ab607b8, msg=0x7f9a8abfef38, c=0x7fffed081380) at action.c:1669 #17 0x00007f9a891f6d0f in prepare_new_uac (t=0x7f9a6de86518, i_req=0x7f9a8abfef38, branch=0, uri=0x7fffed0814d0, path=0x7fffed0814b0, next_hop=0x7f9a8abff1a8, fsocket=0x0, snd_flags=..., fproto=0, flags=2, instance=0x7fffed0814a0, ruid=0x7fffed081490, location_ua=0x7fffed081480) at t_fwd.c:410 #18 0x00007f9a891faded in add_uac (t=0x7f9a6de86518, request=0x7f9a8abfef38, uri=0x7f9a8abff1a8, next_hop=0x7f9a8abff1a8, path=0x7f9a8abff568, proxy=0x0, fsocket=0x0, snd_flags=..., proto=0, flags=2, instance=0x7f9a8abff578, ruid=0x7f9a8abff590, location_ua=0x7f9a8abff5a0) at t_fwd.c:855 #19 0x00007f9a89201ddd in t_forward_nonack (t=0x7f9a6de86518, p_msg=0x7f9a8abfef38, proxy=0x0, proto=0) at t_fwd.c:1723 #20 0x00007f9a891ee374 in t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, proto=0, replicate=0) at t_funcs.c:355 #21 0x00007f9a8922bd89 in _w_t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, force_proto=0) at tm.c:1517 #22 0x00007f9a8922ceee in w_t_relay (p_msg=0x7f9a8abfef38, _foo=0x0, _bar=0x0) at tm.c:1718 #23 0x000000000041d48d in do_action (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1088 #24 0x0000000000429ba3 in run_actions (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1583 #25 0x000000000042a208 in run_actions_safe (h=0x7fffed0848f0, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1648 #26 0x0000000000542994 in rval_get_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, i=0x7fffed082668, rv=0x7f9a8a83eb60, cache=0x0) at rvalue.c:924 #27 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082668, rve=0x7f9a8a83eb58) at rvalue.c:1918 #28 0x0000000000546fc2 in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082af0, rve=0x7f9a8a83f280) at rvalue.c:1926 #29 0x000000000041cf77 in do_action (h=0x7fffed0848f0, a=0x7f9a8a83fbd0, msg=0x7f9a8abfef38) at action.c:1064 #30 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a838c88, msg=0x7f9a8abfef38) at action.c:1583 #31 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996598, msg=0x7f9a8abfef38) at action.c:712 #32 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a9955b8, msg=0x7f9a8abfef38) at action.c:1583 #33 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996a50, msg=0x7f9a8abfef38) at action.c:1079 #34 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a8fa068, msg=0x7f9a8abfef38) at action.c:1583 #35 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:712 #36 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:1583 #37 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830db8, msg=0x7f9a8abfef38) at action.c:1079 #38 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38) at action.c:1583 #39 0x000000000042a2d0 in run_top_route (a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38, c=0x0) at action.c:1669 #40 0x00000000005095b6 in receive_msg ( buf=0xa78b60 "INVITE sip:0123456789@A.B.C.181:5060;transport=UDP;user=phone SIP/2.0\r\nFrom: <sip:0987654321@A.B.C.148:5060>;tag=BN1507883839-1-1507047459-332794206;tran sport=UDP\r\nTo: <sip:0123456789@H.G.F.D"..., len=1119, rcv_info=0x7fffed084be0) at receive.c:216 #41 0x000000000060945a in udp_rcv_loop () at udp_server.c:521 #42 0x00000000004a6e62 in main_loop () at main.c:1629 #43 0x00000000004acfa6 in main (argc=7, argv=0x7fffed085018) at main.c:2581 Regards, Igor.

2 2

Presence WARNING after publish and no NOTIFY
by Yuriy Gorlichenko 23 Oct '17

23 Oct '17

Hi use 5.0.1 with presence for TCP/UDP endpoints presence works PERFECT but for WS socket presence i see next (small cleaned log here with all described bellow https://pastebin.com/8a77ggj0 ): When WS endpoint subscribes - kamailio After kamailio sends 202 to SUBSCRIBE message (see it on client) it sends fast NOTIFY correcly without any trouble, and i see at the kamailio reply from WS endpoint - 200 WS endpoint added to active_watchers table. But when Subscribed endpoint makes for example some call and presence server sends PUBLISH i see: presence [presentity.c:426]: delete_presentity_if_dialog_id_exists(): Presentity already exists - deleting it At the logs and then presence server doesn't send NOTIFY to WS endpoint What is the issue of this? Can not understand why it happens. Can someone to help me understand this? And the dump i see that kamailio send first NOTIFY form UDP port but then translates it to tls for websocket and i see correct NOTIFY at the WS endpoint

2 4

MoS(QoS) measurement in Kamailio based IMS network
by Kranti Kumar 23 Oct '17

23 Oct '17

Hi All, I am using Kamailio IMS VMimage which is provided by Franz. I am successfully established the call between the IMS clients. Here I would like to measure the MoS mean opnion score (QoS) of Calls (Voice quality). Can any one suggest me / guide me how to measure the same in this IMS network. Thank you so much in advance. Regards, -kranti

2 1

how to bridge two landline users
by 赵国杰 23 Oct '17

23 Oct '17

Hello, In a standard sip flow, the call goes like: sip user A --> kamailio --> pstn --> landline user B. However, when user A has a bad internet access, the audio is broken. So what I want is to let sip user A send a invite to kamailio first, then kamailio send invite to user A and B's landline number through pstn, then bridge the two call together. I understand this can be achieved by using FREESWITCH originate and bridge command. I've tried but there's no audio both ways, which really makes me feel stupid of myself. So I'm wondering if this can be done with kamailio? If so, how? Thanks Jesse

2 1

uac_replace_from and header adulterations
by Alex Balashov 23 Oct '17

23 Oct '17

Hello, I've got a bizarre problem caused by bad UA behaviour: UA A ---> Kamailio (P) ---> UA B 1. UA A sends initial INVITE through P to B; 2. Kamailio (P) makes some modifications to the From header using uac_replace_from() and passes along to B. 3. B sends an in-dialog request (e.g. BYE or reinvite) to A through P; in doing so, it modifies the To (formerly From) value slightly, replacing the hostname portion in the To URI with a different value to the one that was received in the From header. 4. Kamailio relays this in-dialog request to A, but with an adulterated/clipped/truncated/grammatically invalid To header now. 5. A responds with 400 Bad Request due to invalid To header. I would not dispute that UA B should not be modifying the remote URI in this manner. But since it does, it gives rise to two questions: 1. Why does Kamailio not simply discard the modified To header and restore the original value stored in the Record-Route rider parameter? Is it because the Record-Route parameter does not contain the original header value, but rather some data complementary to the current header value? Or is it because the UAC code takes a checksum of the original remote URI header value and stores it, and just checks it when restoring on principle? If so, what's the motive for that? 2. Wouldn't it be better behaviour to simply reject a request so malformed, rather than passing it on with a corrupt restored value? If Kamailio can detect that the header has been tampered with, why pass it on? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Help on Video Conference
by Kranti Kumar 23 Oct '17

23 Oct '17

Hello , I’m trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ My configuration details: HSS - 10.0.0.9 (./hss.sh) PCSCF - 10.0.0.10 (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg) ICSCF - 10.0.0.11(kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg) SCSCF - 10.0.0.12 (kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg) host machine: 10.0.0.5 I am able to register mobile clients ( IMSdroid) and make voice/video calls between them successfully. Now I would like to make a conference call among 3 IMSdroid users but its failing. Could you please suggest me the way forward here. I am thinking to integrate Application server (AS) to enable Video Conferences. Could you please guide me how to enable AS in this VM image and configuration details. Is there any possibility to do Video conference using this current setup(VM image). Could you please suggest me. Thank you so much in advance. Regards, -kranti

2 1

Kamailio IMS deployment
by Ramya Y 22 Oct '17

22 Oct '17

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

5 22

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.

2 1

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

3 3

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

3 2

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

19 Oct '17

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

3 4

port forwarding
by Andrzej Kaczmarczyk 19 Oct '17

19 Oct '17

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

2 1

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

19 Oct '17

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2018 takes place 3-4 February 2018 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 4th of February 2018. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 30th of November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM18/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2018/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 3rd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on the 2nd of February 2018 http://wiki.xmpp.org/web/Summit_22 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 2nd of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

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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 ---

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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

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Handle SDP ICE candidates
by Thiago Guimarães 13 Oct '17

13 Oct '17

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

3 4

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

13 Oct '17

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

1 0

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

13 Oct '17

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

3 5

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

13 Oct '17

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

2 1

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

12 Oct '17

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

1 0

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

12 Oct '17

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

2 18

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

12 Oct '17

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

4 4

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

12 Oct '17

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

3 3

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

11 Oct '17

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

3 5

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

11 Oct '17

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

1 0

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

11 Oct '17

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

2 4

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

11 Oct '17

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

2 4

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

11 Oct '17

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

2 1

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

11 Oct '17

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

2 1

Route Not accessable
by anil kumar k 11 Oct '17

11 Oct '17

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

2 1

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

10 Oct '17

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

2 2

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

10 Oct '17

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

1 1

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

10 Oct '17

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

1 0

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

09 Oct '17

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

1 0

Set music on hold
by Fedor 08 Oct '17

08 Oct '17

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

2 1

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

06 Oct '17

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

1 0

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

06 Oct '17

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

1 0

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

06 Oct '17

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

2 1

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

06 Oct '17

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

3 3

SIP -
by Subba Krishna MG 06 Oct '17

06 Oct '17

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

3 5

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

06 Oct '17

I found that the problem is because of this header:

1 1

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

05 Oct '17

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

3 6

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

05 Oct '17

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

3 4

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

04 Oct '17

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

1 0

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

04 Oct '17

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

2 3

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

04 Oct '17

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

1 0

Manipulate From URI username
by Gertjan Wolzak 04 Oct '17

04 Oct '17

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

4 3

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

03 Oct '17

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

1 0

Debian Stretch
by Jurijs Ivolga 03 Oct '17

03 Oct '17

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

5 7

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

02 Oct '17

Hi I added this eventroute for manipulate temprorary entires in my subscriber table but it is not called in my cfg. May be i didn't setted up in scf but didnf found any special params for this. I use kamailio 5.0.3 version: kamailio 5.0.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 09:39:55 Sep 8 2017 with gcc 4.9.2 My registrar and usrloc modules setted up as usual modparam("registrar", "method_filtering", 1) modparam("registrar", "max_expires", 3600) modparam("registrar", "gruu_enabled", 0) modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) thx for advice

2 2

sigcomp usage with kamailio
by Ram 02 Oct '17

02 Oct '17

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

2 1

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

02 Oct '17

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

2 5

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

02 Oct '17

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

2 1

Hello
by Subba Krishna MG 29 Sep '17

29 Sep '17

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

1 0

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

29 Sep '17

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

2 3

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

29 Sep '17

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

2 4

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

28 Sep '17

I have issue with kazoo module call sprintf <https://github.com/kamailio/kamailio/blob/master/src/modules/kazoo/kz_pua.c…> not work properly. body variable contains values like \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"*

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