Re: [SR-Users] memory is not freed in system, while kamailio's statistic shows OK

Hi Daniel! Thanks for the pointing out! Currently re-checking system memory usage and monit behavior. Will report results later (seems that there is nothing to be done on the kamailio's point of view) Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/memory-i

[SR-Users] memory is not freed in system, while kamailio's statistic shows OK

Hi folk! Have a production server - used as webrtc to sip proxy - it is built on kamailio v 4.4.2 - Server handles connections from ~4000 incoming SIP trunks (transport=TLS). Kamailio provides auth for this trunks. - Kamailio receives also connections from Webrtc app. Calls from Webrtc routed to on

Re: [SR-Users] Kamailio + rtpproxy + SRTP

Konstantin M. wrote > I have some problem with SSL+rtpproxy, kamailio successfully decodes SIP, > rtpproxy decodes SRTP frames, all is going to freeswitch unencrypted. FS > transfers the call back, the call is going unencrypted though kamailio, > kamailio encodes to SSL, but rtpproxy DOESN'T encode

Re: [SR-Users] Custom header on reply is removed on t_relay

Hi! check with ngrep/tcpdump, that reply from P1 to P2 indeed consist the header kamailio should no remove any custom headers cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/Custom-header-on-reply-is-removed-on-t-relay-tp150476p150495.html Sent from the Users ma

Re: [SR-Users] Why SDPOPS does not remove attributes in SDP

Hi all these lines are added after rtpengine_manage? if yes,try msg_apply_changes() after rtpengine_manage and before deletion. ! read the description of msg_apply_changes, it may brake something! cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/Why-SDPOPS-does

Re: [SR-Users] HTTP query at the end of a phone call

Hi! Instead of function from utils module, it is better to use newer from here: http://kamailio.org/docs/modules/4.4.x/modules/http_client.html ore here: http://kamailio.org/docs/modules/4.4.x/modules/http_async_client.html hope this is helpful! cheers -- View this message in context: http:/

Re: [SR-Users] 404", "Not Found location"

Hi! While REGISTER processing, lookup(location) is not what you need to call lookup - have to be used, e.g. while INVITE processing, when you need to find location of registered user, to send him INVITE. about name in RURI, if you are talking about REGISTER - it is specific message, and RURI in R

Re: [SR-Users] Push notification question

Hi! To cope with your specific cases try connection-oriented transport (tcp/tls). e.g. if app is closed - the connection is closed as well and the usrloc record will be removed. After this use standard approach (with late forking - tsilo module) imho, it is not a good idea to suspend the call i

Re: [SR-Users] Kamailio setting the reqeust URI

NITESH BANSAL wrote > Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in > SIP INVITE coming from Asterisk looks like this kamailio@.x > >But my objective is to use Kamailio to forward the call to a remote > endpoint. > What header should I put in the SIP INVITE from

Re: [SR-Users] for kamailio registration

Hi! "to see the complete traces" - use ngrep/tcpdump also use kamailio's sipcapture and siptrace modules, that were suggested earlier. cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/for-kamailio-registration-tp146556p146558.html Sent from the Users mailing

Re: [SR-Users] srv_query question

Hi all! any suggestions here? thanks in advance for any idea or thought! cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/srv-query-question-tp146546p146594.html Sent from the Users mailing list archive at Nabble.com. _

Re: [SR-Users] srv_query question

Hi Daniel! Thank you for suggestion! will give it a shot, after getting working result - report here. cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/srv-query-question-tp146546p146600.html Sent from the Users mailing list archive at Nabble.com.

[SR-Users] srv_query question

Hi folk! I am trying to implement srv_query usage in my script (server is based on Kamailio+Asterisk). The task: when outgoing REGISTER comes from Asterisk (outgoing trunk to operator), and R-URI consists domain name, need to store all ip-addresses of remote side (for following use) Currently, th

Re: [SR-Users] htable - banning users

Hi! you need to add one more case in switch ($retcode) condition: -3 (invalid user) - authentication user does not exist look in complete list of negative codes here: http://www.kamailio.org/docs/modules/4.3.x/modules/auth_db.html#auth_db.f.www_authenticate Hope this helps Cheers! -- View t

Re: [SR-Users] rare impossibility to send requests to TLS connected peer

Hi Daniel! Currently I have complete scheme of the issue (and at last time to describe it here): Issue logic (all mentioned functions from tcp_main.c): SIP phone (A-B) connected via TLS, on port 443 to tcp proxy: 192.168.1.24:3487 <--> 192.168.1.85:443 (connect A) 127.0.0.1:64935 <---> 127.0.0.1

Re: [SR-Users] Handling of messages with manipulated Event header.

Hi Christopher! try this function before search_event: http://www.kamailio.org/docs/modules/4.3.x/modules/textopsx.html#textopsx.f.msg_apply_changes It should help BR -- View this message in context: http://sip-router.1086192.n5.nabble.com/Handling-of-messages-with-manipulated-Event-header-t

Re: [SR-Users] Kamailio v4.3 and Freeswitch Stable Latest

Hi! To understand what is happening, you should analyse SIP exchange, use for example: ngrep -d any -qt -W byline "sip" port 5060 you can also use tcpdump with following studying in wireshark. To find out what part of kamailio's configs SIP message processing passed, add logs, and analyse syslog

Re: [SR-Users] rare impossibility to send requests to TLS connected peer

Daniel-Constantin Mierla-6 wrote > Hello, > > so practically all registration appear to come from 127.0.0.1:xyz, where > xyz is some port value. for devices connected to 443, yes it is (kamailio listen 5060/5061, and devices can connect also directly) > My guess is that the connection to the tc

Re: [SR-Users] rare impossibility to send requests to TLS connected peer

Hi Daniel, thanks for quick reply! e.g. 10.100.1.24 - sip phone 10.100.1.85 - server - sip phone is configured to register to 10.100.1.85:443 - tcp proxy listen on 443 - when SIP request comes, tcp proxy open connect to Kamailio 127.0.0.1:5061, and send request there - tcp proxy does not modify s

[SR-Users] rare impossibility to send requests to TLS connected peer

Hi folk! Have a rare issue with tls connections. Have a server with 100+ active tls connections, that are connected to separate TCP proxy, that listen on port 443. TCP proxy send sip traffic to Kamailio via localhost:5061. In this way all connections in Kamailio are considered from Localhost. I

Re: [SR-Users] Kamailio IMS deployment

Hi Franz! Thanks for sharing! Great job, end excellent materials to get started with students (beside my main work I also teach students in Odessa National Academy of Telecommunication). I was looking for some "light" way how to start with real deployment of IMS, not pure theory, and this is just

Re: [SR-Users] tsilo and linphone android connected via TCP

Hi Federico! Many thanks for suggestion, it works! Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/tsilo-and-linphone-android-connected-via-TCP-tp144224p144234.html Sent from the Users mailing list archive at Nabble.com. __

[SR-Users] tsilo and linphone android connected via TCP

Hi folk! sorry for first sending to the wrong list ! Currently playing around with push notification in kamailio, have a bad case. Would appreciate any idea. Scenario (all devices connected via TCP): UA-A - sipphone, UA-B[1] - sipphone UA-B[2] - Linphone android Server - kamailio+asteris

Re: [SR-Users] websocket sometimes log errors after ws connect has closed

additionally: version: kamailio 4.2.5 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, USE_TCP, USE_TLS, 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

[SR-Users] websocket sometimes log errors after ws connect has closed

Hi list! firstly, the described below issue is harmless to us, and is just little bit annoying, but analysing the nature of issue decided to share the results, and hear the opinion of community. Scenario: - WebRTC client connected to Kamailio's websocket - after some time I close tab with WebRTC,

Re: [SR-Users] Kamailio as a SBC

hi! for reply, to pass rtpproxy_manage, look in this function (in your config it is not used, and onreply_route is not passed for replies) http://kamailio.org/docs/modules/4.2.x/modules/tm#tm.f.t_on_reply about segfaults - provide more info also can suggest, try using rtpengine instead of rtppro

Re: [SR-Users] WebRTC no longer supports RTP

Hi! use DTLS-SRTP, to say how to handle it with rtpengine - I think you should provide more info about your setup, and call cases Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/WebRTC-no-longer-supports-RTP-tp143834p143836.html Sent from the Users mailing list

Re: [SR-Users] random incorrect method parsing

Hi Daniel! just additional info about the nature of that "extra zero". Vendor use them as a padding bytes RFC http://www.ietf.org/rfc/rfc0894.txt says: "If necessary, the data field should be padded (with octets of zero) to meet the Ethernet minimum frame size. This padding is not part of

Re: [SR-Users] random incorrect method parsing

Hi Daniel! Thank you for the suggestion, after the deep investigation, looks like the root of the problem found: when device send TCP ACK (before INVITE), in this packet is incapsulated "VSS-Monitoring ethernet trailer, Source Port: 0", example in file in attachment. Looks like kamailio parse it

[SR-Users] random incorrect method parsing

Hi folk! Have a strange issue, and cannot understand what is wrong. Test scheme UA(sip) -> INVITE -> Kamailio The transport protocol used is TCP. The issue is reproduced randomly, in case of wrong INVITE, Kamailio does not parse Method from R-URI and answer "400 CSeq method does not match request

Re: [SR-Users] Media address family

Sergey Okhapkin wrote > What is the best way to find out address family (IPV4 or IPV6) of media in > SDP? > My goal is to reject INVITEs which have different address families in > headers > and SDP, i.e. IPv4 INVITE with IPv6 media address in SDP and vice versa. > > > __

Re: [SR-Users] issue with uac_req_send to websocket peer

the solution is found, the root of the issue was in our proxy daemon, that received sip from outside world, and proxy it to kamailio (kamailio received sip via localhost) My ws users, have such received in location: Received:: sip:127.0.0.1:36986;transport=ws I added in NO

Re: [SR-Users] Kamailio can not read to_tag at BYE request

Hi! did you tried using this PV http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#tt_-_to_tag how do you parsing it now? cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-can-not-read-to-tag-at-BYE-request-tp142928p142959.html Sent from the

[SR-Users] issue with uac_req_send to websocket peer

Hi folk! Have a scenario: UA-A -> REGISTER -> kamailio kamailio make auth, and save(location), after this, before exit, I want to send to UA-A a custom NOTIFY with specific event (this should be done without SUBSCRIBE/PUBLISH/NOTIFY logic) $uac_req(method) = "NOTIFY"; $uac_

Re: [SR-Users] issue with sending "delete" to rtpengine for cancelled branch

sorry for the noise, have misread description in manuals. only needed to call rtpengine_delete when handling 487 from WebRTC to CANCEL closed cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/issue-with-sending-delete-to-rtpengine-for-cancelled-branch-tp142588p

[SR-Users] issue with sending "delete" to rtpengine for cancelled branch

Hi folk! Have an issue, and do not see the obvious way how to solve it, maybe i'm missing something. The call scenario: UA-A (hard phone, no rtpengine) call ->> UA-B (has 2 registration: hard phone (no rtpengine) and WebRTC based web phone (here the rtpengine is called, in branch_route)) When K

Re: [SR-Users] Authentication issue during registration

Hi! Daniel suggested you the right way to understand what is going on. Use something like this (use your appropriate PV instead of mine), and the reason of repeatedly challenging will reveal to you: if (!pv_www_authenticate("$td", "$avp(password)", "0")) { $var(retcode) = $retcode

Re: [SR-Users] during registration nonce expired, after backwards time shift

the pull request prepared, hope it is well formatted (did not have such experience before) Daniel, thank you for spending time on this issue, and help in finding the right way ) -- View this message in context: http://sip-router.1086192.n5.nabble.com/during-registration-nonce-expired-after-bac

Re: [SR-Users] during registration nonce expired, after backwards time shift

Hi! tested with: + /* if current time is less than start time, reset the start time + (e.g., after start, the system clock was set in the past) */ + t=time(0); + if (t < up_since) + up_since = t; + if

Re: [SR-Users] during registration nonce expired, after backwards time shift

on, but then it should recover for that > process. it may actually be even better to test if since >= t and don't > 'return 4' in that case. > > Cheers, > Daniel > On 01/09/15 16:49, Vasiliy Ganchev wrote: >> Hi Daniel! >> >> Any comments/idea a

Re: [SR-Users] during registration nonce expired, after backwards time shift

Hi Daniel! Any comments/idea about my last answer? cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/during-registration-nonce-expired-after-backwards-time-shift-tp140536p140945.html Sent from the Users mailing list archive at Nabble.com. ___

Re: [SR-Users] during registration nonce expired, after backwards time shift

time computed internally as start time >> plus elapsed seconds (counted internally). We can introduce an option to >> use the system time, which may add a bit of delay, but really >> insignificant. I will look into it these days. >> >> Cheers, >> Daniel >> >>

Re: [SR-Users] during registration nonce expired, after backwards time shift

time computed internally as start time >> plus elapsed seconds (counted internally). We can introduce an option to >> use the system time, which may add a bit of delay, but really >> insignificant. I will look into it these days. >> >> Cheers, >> Daniel >> >>

Re: [SR-Users] how to use seturi with varibales in kamailio

Hi! 2. you can use something like this $var(newruri)=newrurivalue; $ru = $var(newruri); if you want to change not complete ruri, take a look here for pv related to R-URI (they are R/W) 3. pseudovariables in To header are read only,

[SR-Users] during registration nonce expired, after backwards time shift

Hi list! (sorry for sending first mail to wrong sub forum) Have a following issue: Server with Kamailio restart with wrong time (why - this is separate part of my investigations), timeshift e.g. for 2 hours in feature. After ntp daemon adjust correct time (move time backwards), Kamailio do not a

Re: [SR-Users] SDP IPv4 has been concatenate - no audio

Loic Chabert wrote > Hello everyone, > > I'm trying put kamailio in front of asterisk server farm. Fow now, 2 > asterisk servers are running and i'm trying to make some basic calls > between two UACc. > > All asterisk servers has been ofuscaded from public internet using > 10.189.122.0/24 network

Re: [SR-Users] Check if user is registered after receiving reply

Dmytro Bogovych wrote > Greetings. > I want to check if user is registered before kamailio will try to > route reply to its user agent. > > IMHO it should be: > > onreply_route[MANAGE_REPLY] { > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) -

Re: [SR-Users] sending INVTE with Digest values

Al S wrote > I am registering my client to kamailo successfully. > client --> Register --> kamailioclient <-- 401 with nonce value <-- > kamailioclient --> Register with nonce and md5 response values --> > kamailioclient <-- 200ok <-- kamailio > However, when the client sends an invite with the sam

Re: [SR-Users] Missing BYE at the far end

Hi Kliment! Where the trace is captured? Can you show the log of kamailio (as I understand in trace the BYE is send to kamailio server from Asterisk-N and Kamailio do not transmit it to client, or something different)? cheers Kliment Toshkov, Netfinity JSC wrote > Dear colleagues, > > > We

Re: [SR-Users] Issue handling SRTP and RTP with rtpproxy and rtpengine

#!endif > > if ($avp(mline) =~ "AVP") > > { > > xlog("L_INFO", "Tenemos RTP "); > > xlog("L_INFO", "Lla

Re: [SR-Users] Issue handling SRTP and RTP with rtpproxy and rtpengine

Alberto Sagredo-2 wrote > ... > I have been able to make SRTP To RTP to Asterisk > > But im not able to call between SRTP extensions, i understand also SRTP to > RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine > trasncode) > > > If you need any more info let me know. >

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

Andrey Utkin wrote > 2015-06-23 18:49 GMT+03:00 Daniel-Constantin Mierla < > miconda@ > >: > > But what sniffer should I try instead of ngrep to have more details > and confidence? Hi! Try to use tcpdump, with following review in Wireshark. This should show you complete picture. Cheers --

Re: [SR-Users] Need information regarding kamilio setup to support IPSec

Hi! I think you can use append_hf_value(hf, hvalue) function in config. Use it in onreply_route that your 200OK is passing. Or what is the question? In the string with parameters everything is constant? or some values are variable? Priyaranjan Nayak wrote > Hi Daniel, > > I am sending Security-C

Re: [SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

Richard Fuchs wrote > On 02/06/15 06:35 AM, Vasiliy Ganchev wrote: >> Richard Fuchs wrote >>> On 29/05/15 11:16 AM, Vasiliy Ganchev wrote: >>>> On May 29, 2015; 3:19pm, Richard Fuchs wrote: >>>>> A good way to start debugging this is to run rtpengi

Re: [SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

Richard Fuchs wrote > On 29/05/15 11:16 AM, Vasiliy Ganchev wrote: >> On May 29, 2015; 3:19pm, Richard Fuchs wrote: >>> A good way to start debugging this is to run rtpengine with log-level 7 >>> and post the full log for such a call. >> Hi Richard! Thanks f

Re: [SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

On May 29, 2015; 3:19pm, Richard Fuchs wrote: > A good way to start debugging this is to run rtpengine with log-level 7 > and post the full log for such a call. Hi Richard! Thanks for answer! Call log written on WS_Kamailio, rtpengine with log-level 7 Call from UA_WS 272 calling to UA_SIP 271 a

Re: [SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

Юрий, Thanks for your answer! Sorry but I couldn't understand how to do "Reverse rtp_manage functions". If it is possible, extend your idea with some more example/explanation? Maybe someone else can help suggest something, or do I need to give more information? Thanks in advance! -- View this

[SR-Users] RTPengine+Kamailio, 200 OK without DTLS fingerprint

Hello list! Have some difficulties with RTPengine: My test scheme is: UA(WS) -> Kamailio (WS proxy) -> Kamailio (SIP REGISTRAR, proxy) -> ASTERISK (media server) -> UA (SIP) Kamailio (WS) use RTPengine. Kamailio (WS proxy) - 4.2.3 Kamailio (SIP REGISTRAR, proxy) - 4.1.4 RTPengine - 3.3.0.0+0~mr3.8

Re: [SR-Users] Kamailio proxy for far-end nat traversal

Looks like UAb (3G) do not receive the INVITE (or makes no answer for some reason). Can you check if UAb is receiving the INVITE? LinphoneAndroid that you use as a User-Agent on UAb has to log something. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-proxy-f