- sr-users - kamailio.org

Kamailio World Updates
by Daniel-Constantin Mierla 25 Feb '13

25 Feb '13

Hello, the next group of accepted talks at Kamailio World Conference will be published in few days, keep an eye on: - http://conference.kamailio.com/k01/schedule/ Till then, here is what happened lately around the project. Olle E. Johansson spent the week at SIPit, making sure the upcoming version 4.0.0 is ready for release: - http://www.kamailio.org/w/2013/02/sipit30/ I conducted the Kamailio Development Workshop, with a great group of participants, first results being visible as contribution to app_lua for review on tracker. Today is schedule to release Kamailio v3.3.4, a new minor version out of latest stable branch 3.3. This week on Wednesday, development branch will be unfrozen, lots of new code is already waiting to be merged there (to be part of future version 4.1.0). If you are in Berlin this week, participating at Call Center World (http://www.callcenterworld.de/en/), ping me to meet, I will be glad to catch up with what is happening on real time communications market. Do not forget that there are open positions to work with VoIP/SIP and Kamailio, just in case you look for new challenges in your career, check: - http://conference.kamailio.com/k01/career-jobs/ It is less than two months till Kamailio World Conference, the schedule is almost filled up, if you haven't registered yet, it is time to hurry up, don't miss the chance to meet many professionals working with large real time communications systems or solutions, from Kamailio or different other projects: - http://conference.kamailio.com/k01/registration/ At the end, many thanks to those sponsoring the event, helping to cover the logistics to ensure a great event. If you consider sponsoring, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Kamailio 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)
by Rowie 24 Feb '13

24 Feb '13

Hi, We are having an issue where a phone (snom in particular) cannot make a call through Asterisk. It just hangup and does not allow the call to go through. I am including a a sip trace on this thread to show what is happening within the call. Please see below: Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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SOS!! Kamailio 3.1.x Load Balancer doesn't work
by chanea 24 Feb '13

24 Feb '13

Dear All, I am trying to use Kamailio 3.1.x server (called LB here) as the Load Balancer for two other Kamailio server (called server01 and server02 here) . Client A -----> Kamailio LB -----> Kamailio server01 Client B -----> Kamailio LB -----> Kamailio server01 If I dont use LB here , and letting Client A and B connect to Kamailio server01 directly , all the communication is fine. While using Kamailio LB ,without luck, although the sip client A and B get 200 from server01 for Register, they cannot reach each other for "Message" and "Invite". It will be great appreciated if anyone can give a hint. my config for Kamailio LB is as below : ========================================== .... debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 # for more info: sip_router -h # ------------------ module loading ---------------------------------- mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "dispatcher.so" loadmodule "db_mysql.so" # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("dispatcher", "db_url","mysql://.........@........") modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; # if (!method=="REGISTER") # record_route(); ds_select_dst("1","4"); forward(); exit; } .. ========================================== best regards, chanea

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SIPit 30 summary
by Olle E. Johansson 22 Feb '13

22 Feb '13

Hi! I've spent five days at SIPit 30 testing Kamailio on a tiny Asus EEE-netbook. It's been great, and apart from the bug hitting me while using outbound, we've done great in all tests. During the proxy doom test today, the netbook processed way over 50.000 messages in a single INVITE transaction that spiralled and looped and behaved in funny ways, and the CPU still had a lot of power to do other things. The hub we used was the limiting factor. Kamailio 4.0.0 beta was involved in all multiparty tests, from TLS and RTCweb to forking, spiralling and proxy doom. We proved that we have a solid product that can handle all kind of scenarious, even TCP overflow attacks. Thanks for all the help over jabber, IRC and other means! Special thank you to Fred Posner who contributed a Big Fred Cookie! Have a great weekend! /O

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Path and DNS SRV
by Charles Chance 22 Feb '13

22 Feb '13

Hi All, Given a situation where there are multiple registrars, behind a single load balancer, sharing the same location table in DB, I understand that Path can be used to ensure everything works nicely with clients behind NAT. So, if UA1 is registered on P1, UA2 on P2, and UA1 calls UA2... P1 performs lookup, sees path and sends via P2. What about situations where the distribution between registrars is being handled client-side through DNS SRV, not a load balancer? Path cannot be used then, right? As I understand it, in this instance add_path() will do nothing, is this correct? Thanks in advance, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

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Weird out-of-memory message on start/stop
by Andreas Granig 21 Feb '13

21 Feb '13

Hi, Maybe some of you have encountered this funny error message when starting and even killing kamailio on a Debian 6 with 2.6.x and 3.x kernels and 2.11.3-4: #------------------------- # /usr/sbin/kamailio -f /etc/kamailio/proxy/kamailio.cfg -P /var/run/kamailio/kamailio.proxy.pid -m 4 -M 16 -u kamailio -g kamailio loading modules under /usr/lib/kamailio/modules_k:/usr/lib/kamailio/modules Listening on udp: 127.0.0.1:5062 tcp: 127.0.0.1:5062 Aliases: Message from syslogd@sp2 at Feb 19 11:15:42 ... t of memory [6720] # killall kamailio Message from syslogd@sp2 at Feb 19 11:17:51 ... t of memory [6720] #------------------------- The PID reported here is the kamailio attendant process. It seems to be somehow related to the shared memory. The interesting part about that is that kamailio really starts just fine, and there is plenty (>5GB) of free memory available. It doesn't seem to influence the behaviour of kamailio at all as it's working perfectly fine, so for now it's just annoying, but I'm really curious I've tried to fiddle with mem_join, but it doesn't matter whether it's on or off. Kamailio version is this: # kamailio -V version: kamailio 3.3.2 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 19:09:17 Jan 21 2013 with gcc 4.4.5 Has anyone seen this too? What's the cause of that? Andreas

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SDP c= address as a key for a key-value lookup?
by Hegner, Travis 21 Feb '13

21 Feb '13

Hi All, I am working on a kamailio proxy with a very specific goal I need to accomplish. What I would like to do is a key-value lookup, utilizing the SDP's media contact (c= line) IP as a key, and have a function return a value for that key. It appears that I can possibly accomplish this with the AVPops module, combined with DB_TEXT (maybe DB_MYSQL in the future) and a custom table with a custom query. The only thing I seem to be missing is the ability to extract the c= IP from the SDP, and store it as some kind of variable with which to do the lookup. I've looked into the iptrtpproxy module as it seems to have a select: @iptrtpproxy.sdp_ip that could work, but that module is extreme overkill for my purposes, and I am running into some issues getting it built for my platform (Ubuntu server 12.04). Also the documentation for that select states: "Return first rewritten IP provided at SDP c= line." However, I'm looking for the original sdp_ip, and have some concern that this won't return an IP that has not yet been re-written. My environment has a very specific 1:1 ratio of private network, to static public IP mappings. I currently have a successful OpenSER (1.3.2) deployment within this environment (running 5 years), which uses a hacked version of the old "mangler" module, with the function sdp_mangle_ip(<network>, <ip>). I could do something similar using textops->search_body() and nathelper->fix_nated_sdp(), but the risk of a typo or syntax error in the config file every time I have to make a change or add a network is fairly high. Less importantly, there is some extra overhead as the proxy will have to test for every network on every SDP looking for a match (this is what the current deployment does). Eventually I'd like to add a simple frontend to add/edit networks on the fly, so that no one is ever touching the config. What would you folks recommend for this? Should I look at building a custom module which only grabs the c= IP and stores it into an AVP? Am I not seeing another module that will expose that information for me already? Would it simply be easier to work on getting iptrtpproxy built, and utilizing it only for the purpose of extracting the information I need from the SDP? Any insight would be most greatly appreciated. Thanks, Travis Hegner ________________________________ The information contained in this communication is confidential and is intended only for the use of the named recipient. Unauthorized use, disclosure, or copying is strictly prohibited and may be unlawful. If you have received this communication in error, you should know that you are bound to confidentiality, and should please immediately notify the sender or our IT Department at 866.459.4599.

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Difference between shared and script variable
by Mino Haluz 21 Feb '13

21 Feb '13

Hi, what is the difference between shared and script variable ? Thanks Mino

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Re building command
by Techie Sup 20 Feb '13

20 Feb '13

Hi all, Sorry for this simple question. 1) I have made some changes to main.c file and want rebuild kamailio so that changes take Éffect, what is the command to rebuild? Thanks Selvam

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using dlg_manage before or after?
by Uri Shacked 20 Feb '13

20 Feb '13

Hi, I am trying to change the timeout of a dialog based on the $avp() I set on the modeparam. While testing, I notice that the change does not apply after I am using dlg_manage, only if i changed it before. Can anyone explain why is that? and is it possible to change it after doing dkg_manage? Thanks, URi

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mojtaba jahanbakhsh
by mojtaba jahanbakhsh 20 Feb '13

20 Feb '13

http://www.firstchoiceforyou.com/ecj/xnyfbj4f=4dbl9cu52nkdz

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Strip All but numbers/*/#
by Ali Jawad 20 Feb '13

20 Feb '13

Hi I am using the below to strip "-" from destination numbers, it does work just fine "I replace - with nothing", what i need to do now is to strip all characters except 0-9 * and #. Is there anyway to do it using the following configuration part if($rU =~ ".*-.*" ) { $rU = $(rU{re.subst,/-//g}); } Regards

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Re: [SR-Users] kamctl restart and mediaproxy
by Victor Seva 19 Feb '13

19 Feb '13

We use mediaproxy and kamailio and using the init.d script from the Debian package to restart kamailio has no effect on mediaproxy process. 2013/2/8 Eduardo Lejarreta <lejarreta.e(a)sarenet.es>: > Good morning Daniel. > > I send you the "messages" log which is a bit filtered. The flow is: > > Start a call, engage_mediaproxy on ports (50156, 50157, 50158, 50159) and > then "kamctl restart". As you see it seems that an instruction from kamailio > tells mediaproxy to drop the ports. > We have tried to script around "kamctl" to avoid sending orders to > mediaproxy, killing the "media-dispatcher" before Kamailio and then > restarting again (media-dispatcher) and this works OK, but seems not be a > reasonable way to fix it. > > ...... > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50156 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50157 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50158 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50159 > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3461]: NOTICE: <core> > [main.c:706]: Thank you for flying kamailio!!! > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3561]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3558]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3560]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3553]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3557]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3549]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3552]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3547]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3545]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3546]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3541]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3538]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3536]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3526]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3492]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3528]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3521]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3530]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3495]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3532]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3493]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3539]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3523]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3534]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3489]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3496]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3488]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3525]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3487]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3498]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3484]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3500]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3482]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3501]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3479]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3499]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3474]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3476]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3475]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3477]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3491]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3480]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3490]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3486]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3483]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50156 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50157 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50158 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50159 Closed) > Feb 8 11:05:08 kamailio-dev kamailio: INFO: <core> [tcp_main.c:4727]: > init_tcp: using epoll_lt as the io watch method (auto detected) > Feb 8 11:05:08 kamailio-dev /sbin/kamailio[3761]: INFO: usrloc > [hslot.c:53]: locks array size 512 > ...... > > Thanks and regards. > -- > Eduardo Lejarreta. > > -----Mensaje original----- > De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Enviado el: viernes, 08 de febrero de 2013 9:38 > Para: lejarreta.e(a)sarenet.es; SIP Router - Kamailio (OpenSER) and SIP > Express Router (SER) - Users Mailing List > Asunto: Re: [SR-Users] kamctl restart and mediaproxy > > Hello, > > not using mediaproxy, but have you looked at its logs to see if prints some > messages? > > Cheers, > Daniel > > On 2/7/13 5:03 PM, Eduardo Lejarreta Langarica SARENET wrote: >> Good evening again. >> >> Kamailio 3.2.2 and mediaproxy 2.5.2 >> >> An long-time issue for us: >> >> kamctl restart stops mediaproxy audio. >> >> is there anyway to avoid this? >> >> Any idea will be helpful. >> >> Thanks and best regards. >> ---------------------------------------------------------------------- >> ------------------------------------------------------------- >> Eduardo Lejarreta >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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branching for v4.4.x and unfreezing development
by Daniel-Constantin Mierla 19 Feb '13

19 Feb '13

Hello, it is time to sketch the road to the next major release, my plan is: - create branch 4.4 on Wednesday, February 27, 2013 - that will unfreeze development, so personal branches and contributions kept in external repositories can be merged in master branch - consider to release v4.4.0 in first or second week of March Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

Nonce secret and expiry value
by Andreas Granig 19 Feb '13

19 Feb '13

Hi guys, Nonces have a default expiry value of 300 seconds. Does this still apply if the "secret" parameter is set? So can you use both "secret" and "nonce_expire" value together, taking for granted that the time is in sync over multiple proxies? Andreas

2 2

2 kamailio servers as proxy
by Andrei Rotaras 19 Feb '13

19 Feb '13

Hello, We have this situation: 2 servers, each with his own "location". First user is registered on one server and second user on other server. If we receive an invite for a user that is on other server's location, second server doesn't react on hostrewrite(dynamic routing,forward, append_branch, t_replicate, etc) from first server. Can you please advise us in what direction to look? Our kamailio version: kamailio-3.3.1-1.2.x86_64. (installed from rpm) Thank you! Best regards, Andrei Rotaras NOC Systems Engineer StarNet Moldova Phone: +(373-22) 844-770 Fax: +(373-22) 844-445 Mob: +(373-79) 700-552 E-mail: arotaras(a)starnet.md<mailto:arotaras@starnet.md> Address: 4th Floor, Calea Iesilor 10, Chisinau, Republic of Moldova Web: http://www.starnet.md<http://www.starnet.md/> [Description: Description: Description: starnet] This email message is confidential and may contain privileged information. If you are not the above named addressee, it may be unlawful for you to read, copy, distribute, disclose or otherwise use the information in this email message. If you are not the intended recipient of this email message please telephone one of the above indicated numbers immediately and delete this message.

2 1

Any substitute or replace function for the request-/response line of SIP messages available?
by Klaus Feichtinger 18 Feb '13

18 Feb '13

Hi, I have a very specific situation that the implementation of a customer SIP user agent is absolutely not standards conform. Therefore, I use Kamailio as a SIP protocol adapter between the non-standard conform user agent(s) and the typical RFC3261 compliant user agents. Most conversions are working fine. But in a very specific scenario I have to change/convert a SIP response message to a SIP request message and change the SIP method´s name. This conversion is no problem for the SIP header fields. However, I haven't found a way for adapting the response-/request line of a SIP message. This line seems to be protected. The described situation of the perverse conversion I try to realize is following (please do not ask "why?" or think I'm mad...): (mad) UA1 == INVITE ==> SIP-Prot-Adapter(Kamailio) == INVITE ==> (typ.)UA2 UA1 <== BYE == SIP-Prot-Adapter(Kamailio) <== 488 == (typ.)UA2 UA1 == BYE ==> SIP-Prot-Adapter(Kamailio) == ACK ==> (typ.)UA2 I use the "textops" module for substitution and/or replacement in SIP header fields. But the "subst" and "replace" functions cannot be used for the request-/response line on top of the SIP messages. I was a little bit irritated, as the "search" function is still working on these lines, but not the subst/replace function. Creating a new SIP message with the UAC module (as alternative) is not okay, as this new message does not use the same values for Call-ID, From-tag a.s.o. and these values can't be changed before sending the message. Does anybody know a way / function / method for manipulating the request-/response line of a message? It would be perfect using a function á la "sed"/subst. Thanks Klaus P.S. I tested with kamailio-3.3.3

2 3

Kamailio and Ubuntu Upstart script
by Moacir Ferreira 17 Feb '13

17 Feb '13

As I posted here before, Ubuntu has moved to Upstart as the main engine to start daemons. As a result, Kamailio will fail to start on boot time because it starts before MySQL that has already been moved to the new Upstart daemon starting fashion. I am struggling to get some help from the Ubuntu community explaining how we can match the previous init script behavior when using Upstart but I got no success till now. Anyway, I have created an Upstart script for Kamailio that does most everything that the old script use to do. However, in this script I am making sure MySQL starts before Kamailio. If you have another SQL engine you must change the script to match your SQL engine by changing the “start on started” statement on the script. To use it you have to: A – Create a kamailio.conf file at “/etc/init/” with the contents of the kamailio.conf script in attach, that is the new Upstart stile init script. Please notice that I just adapted the existing script, no news… B – Create a symbolic link at /etc/init.d/ pointing to this kamailio.conf at /etc/init/. C – If you want, you can still have a /etc/default/kamailio file like it used to be and the script will behave like it used to behave when using the default file. Hopefully someone more “influent” at the Ubuntu community may get some help on how to adapt their “start on started” Upstart stanza to behave like we need for multiple possible SQL engines. Cheers! Moacir

3 2

app_python rewrite_ruri - BUG: do_action bad set*()
by V Tone 15 Feb '13

15 Feb '13

Hi, I'm having problems getting started with the app_python module. I'm getting the following in the logs: 5(8890) : <core> [action.c:785]: BUG: do_action: bad set*() type 14 5(8890) ERROR: <core> [action.c:1568]: run action error at: :0 5(8890) ERROR: app_python [python_msgobj.c:115]: Error in do_action 5(8890) ERROR: app_python [python_support.c:38]: python_exec2: Unhandled exception in the Python code: 5(8890) ERROR: app_python [python_support.c:78]: RuntimeError: Error in do_action My Python module is very basic: class Router(object): def child_init(self, child_id): return 0 def route(self, msg): msg.rewrite_ruri('sip:1000@1.2.3.4:5060') return 0 def mod_init(): instance = Router() return instance I've tested this in Kamailio 3.2 and 3.3. Can someone point me to what I'm doing wrong? Many thanks in advance, vtone

2 4

siptrace/sipcapture and MSRP
by Peter Dunkley 15 Feb '13

15 Feb '13

Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter

3 7

Warning - What does this mean?
by Krishna Kurapati 15 Feb '13

15 Feb '13

I am seeing this error lately. /usr/local/sbin/kamailio[28559]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction What does this mean? Krish Kura

1 0

Dialog is not cleaned up after failed calls and BYEs.
by Aft nix 15 Feb '13

15 Feb '13

Hi, I'm trying to limit concurrent calls through kamailio using following script: #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$fd", "$var(100)"); if ($var(100) >= 100 ) { xdbg("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); } #!endif But the dialogs are not cleaned up after the call ends or a failed call happens. For example after two failed calls : [root@new tls]# kamctl fifo profile_get_size callquota profile:: name=callquota value= count=4 and the dialog list : [root@new tls]# kamctl fifo dlg_list dialog:: hash=896:11654 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11755 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=896:11655 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11755 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=896:11657 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11756 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=1532:8089 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: povqQaC.sRhKvmUaLzzR83bRXxouoNy8 from_uri:: sip:123@205.164.40.150 from_tag:: qXQll3QXrff.t27FHY2hucPfJ7K8Pfrs caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 9232 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: You can see there are multiple entries. And these dialog does not get cleaned up. The data is collected when there was no call(pending or connected). My kamailio version is : [root@new tls]# kamailio -V version: kamailio 4.0.0-pre0 (x86_64/linux) 83eafc flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 83eafc compiled on 18:34:58 Jan 17 2013 with gcc 4.4.6 Any suggestion?(It seems dialog module is not working as it should be. I've also tried calling unset_profile() forcefully in failure route with no luck) -- -aft

1 0

Registration and TCP connection
by Krishna Kurapati 15 Feb '13

15 Feb '13

Hi, When client connects to server over TCP/TLS and REGISTERs two things that can happen. 1. Client can disconnect the connection when the App is killed abruptly and the client did not get a chance to send unregiter. 2. CRLF KA did not arrive due to network failures in the middle. Is there a way to remove contact from usrloc table. This can spare server from trying to send message to a client who is no longer connected to the server. It would be ideal if this can be done automatically with a module param in tcp or registrar. Any suggestion on how to tie TCP connection with usrloc? Thanks Krish Kura

2 1

kamailio - simple presence over port 5061
by David Thomson 15 Feb '13

15 Feb '13

I was wondering if simple presence can work over port 5061 or if it always goes over 5060. my client (jitsi) is configured to work over port 5061, xcap is configured to work over port 5061 but I am still seeing some messages flowing over port 5060. any ideas as to what is going on? ttyl,Dave

2 1

[Fwd: Re: Deleting the expired contacts using the Rtimer module]
by kiran bhosale 15 Feb '13

15 Feb '13

he array values that we are modifying and the function that we call periodically are from our module . to exemplify i have put the code below .our requirement is that we want to delete the expired contacts without relying on usrloc module as "we have not used any DB for contact storage" but developed our own module for storing them in a file.even we had installed the SIGALRM handler in our module for decrementing the contacts but the values of array (that stores the expires) elements were zero. we thought of workaround to store the expires values to file and decrement them periodically. but frequent I/O is very consuming. whats the solution for deleting the expired contacts . #define DB_PATH "/usr/local/etc/kamailio/dbtext/syntel" #define DB_PATH1 "/usr/local/etc/kamailio/dbtext/temp" #define MAX_BUFFER 128 #define MAX_USERS 128 typedef struct { int max_bindings; int syntel_index; int expires[8]; char ip_addr[8][16]; } Syntel_users; Syntel_users syntel_database[16]={{0}}; //this is function used for storing the contacts int write_to_file (char *uri, char *user, int locator,int expires) { int Retval = 0; int fd; int bytes; char ip[13]; strncpy (ip, &uri[9], 12); ip[12] = '\0'; fd = open (DB_PATH, O_CREAT | O_RDWR | O_APPEND, 0777); if (syntel_database[locator].max_bindings < 8) { strcpy (syntel_database[locator].ip_addr[syntel_database[locator].max_bindings], uri); syntel_database[locator].expires[syntel_database[locator].max_bindings]=expires; syntel_database[locator].max_bindings++; bytes = write (fd, uri, strlen (uri)); Retval = 1; } return Retval; } //AND THIS IS FUNCTION WE CALL FROM kamailio.conf WITH RTIMER MODULE AFTER INTERVAL OF 1 SECOND. int syntel_delete (void) { int counter; int j; for (counter = 0; counter < 16; counter++) { for (j = 0; j < syntel_database[counter].max_bindings; j++) { if(syntel_database[counter].expires[j]<=0) { memset(syntel_database[counter].ip_addr[j],0,strlen(syntel_database[counter].ip_addr[j])); syntel_database[counter].max_bindings--; } else syntel_database[counter].expires[j]--; } } return 0; } BUT WHAT THIS FUNCTION GETS IS ZERO VALUES FOR EXPIRES AND NOT MODIFIED IN SYNTEL_WRITE_TO_FILE HOW TO DECREMENT THESE VALUES PERIODICALLY THEN ??

2 3

Fwd:
by Maurice Vistan 15 Feb '13

15 Feb '13

http://www.hotelgloriachiancianoterme.it/ako7ed.php?s=lf

1 0

How to rewrite "tu" in CANCEL message ?
by Julia 14 Feb '13

14 Feb '13

Hello all, Is there any way to rewrite "tu" in CANCEL message in 3.2.1? We use remove_hf / insert_hf for INVITE's "tu" modification before t_relay. It works fine, but it doesn't work for CANCEL (INVITE re-parsing is disabled). Ths,

8 19

Openfire and lync 2010 integration
by Vinay V 13 Feb '13

13 Feb '13

Hi All, I am in process of integrating lync 2010 with openfire xmpp server. Reading some of the blogs on kamailio found that it can be done using kamailio. Hence request if some body has done this please your inputs. Regards, Vinay

1 1

Re: [SR-Users] kamctl restart and mediaproxy
by Mino Haluz 13 Feb '13

13 Feb '13

You have to figure it somehow with adjusting the python code of media-dispatcher. We have developed special "/etc/init.d/kamailio grace" command, which waits for the moment, when there are no calls, so it can be restarted. It queries mediaproxy every second. In future we will not use mediaproxy anymore - it is not supported in latest version either. rtpproxy gives us better performance results. On Tue, Feb 12, 2013 at 2:44 PM, Eduardo Lejarreta <lejarreta.e(a)sarenet.es>wrote: > Good evening. > > Launching "mediaproxy-dispatcher" and "mediaproxy-relay" on the foreground > and with "DEBUG" we can see: > > mediaproxy-dispatcher log: > ------------------------------------------------------------------> Kamctl > restart > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Issuing "remove" command to relay at A.B.C.D > debug: Got statistics: {xxxxxxxxxxxxxxxxxxx} > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > > mediaproxy-relay log: > ------------------------------------------------------------------> Kamctl > restart > debug: removing session xxxxxxxxxxxxxxxxxxx > (Port 50000 Closed) > (Port 50001 Closed) > (Port 50002 Closed) > (Port 50003 Closed) > > So "mediaproxy-dispatcher" doesn't receive any command from Kamailio, he > just detects " Connection was closed cleanly" and then issues a "remove" > command to "mediaproxy-relay". > > Does anybody know any way to avoid this? > > Thanks and regards. > -- > Eduardo Lejarreta. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 1

hadnling the IP extension calls freeing the PBX channel
by kiran bhosale 13 Feb '13

13 Feb '13

hello , w have PBX gateway along with kamailio. all calls are routed to pbx extensions. but if it happens to be an IP extrension we want the kamailio to handle the speech without keeping the PBX channel busy. if its an analog extension then this channel can be busy but if its IP extension kamailio should handle the speech itself. how is this possible ?? thanks in advance.

1 0

pseudovariable values written from app_lua sometimes appear incorrect
by elactrum＠jamailca.com 12 Feb '13

12 Feb '13

Hi, I am using app_lua to frequently store some data in htable and avp entries, using sr.pv.sets. When I read the values using sr.pv.get immediately after writing them, they seem correct. However, when I read them later from Lua, I have noticed that sometimes the value is incorrect or nil. Am I doing something wrong? The following code uses rtimer to duplicate the problem on startup, printing the errors to syslog. Tested on version: kamailio 3.3.1 (i386/linux) 2d0f6f ############### # kamailio.cfg ############### loadmodule "pv.so" loadmodule "xlog.so" loadmodule "htable.so" loadmodule "rtimer.so" loadmodule "app_lua" modparam("htable", "htable", "status=>size=14;autoexpire=0;") modparam("rtimer", "timer", "name=ta;interval=10;mode=1") modparam("rtimer", "exec", "timer=ta;route=LUA_TEST") # the following path may need to change depending on filesystem setup modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") route { route(LUA_TEST); } route[LUA_TEST] { $var(lua_test_key) = $(pp{s.toupper}) + "::"; lua_runstring("htable_test([[status=>$var(lua_test_key)]])"); lua_runstring("avp_test([[status=>$var(lua_test_key)]])"); } ---------------- -- kamailio.lua ---------------- -- Test of sr.pv operations in Lua function htable_test (namekey) local pvar_prefix = "$sht("..namekey pvar_test(pvar_prefix) end function avp_test ( name ) local pvar_prefix = "$avp("..name pvar_test(pvar_prefix) end function pvar_test ( pvar_prefix ) for value = 10000, 10050 do local pvar = pvar_prefix..value..")" --sr.log("info", "Lua attempting to set '"..pvar.."' to value '"..value.."'") sr.pv.sets(pvar, value) end for value = 10000, 10050 do -- now we check what was stored local pvar = pvar_prefix..value..")" local checkvar = sr.pv.get(pvar) if ( tostring(checkvar) ~= tostring(value) ) then sr.err(string.format("CHECK: Lua checking pvar '%s' and got value '%s' which is NOT '%s'", tostring(pvar), tostring(checkvar), tostring(value) )) end end end -- end of kamailio.lua Thanks, -Andrew

2 6

DNS error
by Vitaliy Aleksandrov 12 Feb '13

12 Feb '13

Hi, all Have a question about failure_routes. As I remember if kamailio statefuly sent a request and didn't receive a reply script writer can handle this situation at a failure_route[] block. But if kamailio can't resolve a destination from R-URI it sends back 478 reply and finishes current transaction without reaching a failure_route[]. Is it possible to process such a request in a failure_route[] ? Additional details: version: kamailio-3.2.4 t_on_failure() is called before t_relay() I have checked with REGISTER and INVITE transactions.

1 0

Re: [SR-Users] sr-users Digest, Vol 93, Issue 13
by madhumanjusha＠integramicro.com 12 Feb '13

12 Feb '13

Yes,Yes I did it the way tm and mediaproxy modules are added... when i check /usr/local/lib/kamailio/modules I found my module and $stat module time is effected. SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> > seems that you want to create your custom module. have you checked the > makefile of your module. in order to develop the custom module just copy > the existing module and make relevant changes there. this is genuine way . > for this u need to include makefile.lst ,Makefile in your module. so best > way is to copy the smallest module (e.g.) pdb module and make changes. > On Fri, February 8, 2013 3:36 am, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Reg. shared library linking during compiling time > (madhumanjusha(a)integramicro.com) > 2. Re: Reg. shared library linking during compiling time > (kiran bhosale) > 3. Re: Multiple radius responses on a missed call > (Daniel-Constantin Mierla) > 4. Re: Reg. shared library linking during compiling time > (Daniel-Constantin Mierla) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Fri, 8 Feb 2013 01:46:58 -0500 (EST) > From: madhumanjusha(a)integramicro.com > Subject: [SR-Users] Reg. shared library linking during compiling time > To: sr-users(a)lists.sip-router.org > Message-ID: > <38400.61.8.152.138.1360306018.squirrel(a)mail.integramicro.com> > Content-Type: text/plain;charset=iso-8859-1 > > > > Hello All, > Please find below steps...and kindly let me know the error solution. > > > STEP 1:: > Under modules I created a directory name b2bentities > /root/kamailio-3.3.2/modules/b2bentities > $ls > b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o > dialog.h librpath.lst makecfg.lst Makefile > > In b2b_server.h > #ifndef SERVER_H > #define SERVER_H > #endif > > > int b2b_dlg_reply(); > > > > In b2b_server.c I wrote following code :: > > > int b2b_dlg_reply() { > LOG(L_INFO,"Hello am in b2bua module\n"); > return 0; } > > > Also,I added below code to the same file as per steps given in kamailio > server's developer's guide. In developer's guide,they told to initialize > parameters as per module's requirement but i made them all empty and > null... for example static int mod_init(void) { > LOG(L_INFO,"IN mod_init\n"); > return 0; } > > > static cmd_export_t commands[] = { }; > > > > static param_export_t params[]={ }; > > > struct module_exports exports = { "b2bentities", // module name > NULL, // dlopen flags > commands, // exported functions params, // exported parameters > NULL, // exported statistics > NULL, // exported MI functions > NULL, // exported pseudo-variables > NULL, // extra processes > mod_init, // module init function (before fork. kids will inherit) > NULL, // reply processing function > NULL, // destroy function > child_init // child init function }; > > > > STEP 2:: > In kamailio core > file : In receive.c Method Name : In receive_msg > > > i did this way... > > In receive.c add follow piece of code:: > > > #include "modules/b2bentities/b2b_server.h" > > > int receive_msg(....) { > /*my code*/ > int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; > if(msg->REQ_METHOD!=METHOD_REGISTER) { > > > b2b_ret=(*ptr)(); if(b2b_ret!=0) { > LOG(L_INFO,"B2B Dialogue reply function failed\n"); > exit(1); } > else { > LOG(L_INFO,"B2B Dialogue reply function success\n"); > exit(1); } > > > } > > > > } > > > STEP 3:: > I even added kamailio.cfg load module "b2bentities.so" > > > > > STEP 4:: > Final step,when I do make all..... > its giving me the following error :: [root@kamserver kamailio-3.3.2]# make > all CC (gcc) [kamailio] receive.o > LD (gcc) [kamailio] kamailio > receive.o: In function `receive_msg': > /root/kamailio-3.3.2/receive.c:163: undefined reference to `b2b_dlg_reply' > collect2: ld returned 1 exit status > make: *** [kamailio] Error 1 > > > > > My doubt: > Where should my b2bentities.so should be linked during compiling time? > Please tell me the steps for it. > > > Thanks & Regards, > Manjusha A. > Integra Micro Software Services (P) Ltd. > > > > > > > > > > > ------------------------------ > > > Message: 2 > Date: Fri, 08 Feb 2013 12:36:36 +0530 > From: kiran bhosale <kbhosale(a)synteltelecom.com> > Subject: Re: [SR-Users] Reg. shared library linking during compiling > time To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114A3FC.9000902(a)synteltelecom.com> > Content-Type: text/plain; charset=us-ascii; format=flowed > > > madhumanjusha(a)integramicro.com wrote: >> Hello All, >> Please find below steps...and kindly let me know the error solution. >> >> >> STEP 1:: >> Under modules I created a directory name b2bentities >> /root/kamailio-3.3.2/modules/b2bentities >> $ls >> b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o >> dialog.h librpath.lst makecfg.lst Makefile >> >> In b2b_server.h >> #ifndef SERVER_H >> #define SERVER_H >> #endif >> >> >> int b2b_dlg_reply(); >> >> >> >> In b2b_server.c I wrote following code :: >> >> >> int b2b_dlg_reply() { >> LOG(L_INFO,"Hello am in b2bua module\n"); >> return 0; } >> >> >> Also,I added below code to the same file as per steps given in kamailio >> server's developer's guide. In developer's guide,they told to >> initialize parameters as per module's requirement but i made them all >> empty and null... for example static int mod_init(void) { >> LOG(L_INFO,"IN mod_init\n"); >> return 0; } >> >> >> static cmd_export_t commands[] = { }; >> >> >> >> static param_export_t params[]={ }; >> >> >> struct module_exports exports = { "b2bentities", // module name >> NULL, // dlopen flags >> commands, // exported functions params, // exported >> parameters NULL, // exported statistics >> NULL, // exported MI functions >> NULL, // exported pseudo-variables >> NULL, // extra processes >> mod_init, // module init function (before fork. kids will >> inherit) NULL, // reply processing function >> NULL, // destroy function >> child_init // child init function }; >> >> >> >> STEP 2:: >> In kamailio core >> file : In receive.c Method Name : In receive_msg >> >> >> i did this way... >> >> In receive.c add follow piece of code:: >> >> >> #include "modules/b2bentities/b2b_server.h" >> >> >> int receive_msg(....) { >> /*my code*/ >> int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; >> if(msg->REQ_METHOD!=METHOD_REGISTER) { >> >> >> b2b_ret=(*ptr)(); if(b2b_ret!=0) { >> LOG(L_INFO,"B2B Dialogue reply function failed\n"); >> exit(1); } >> else { >> LOG(L_INFO,"B2B Dialogue reply function success\n"); >> exit(1); } >> >> >> } >> >> >> >> } >> >> >> STEP 3:: >> I even added kamailio.cfg load module "b2bentities.so" >> >> >> >> >> STEP 4:: >> Final step,when I do make all..... >> its giving me the following error :: [root@kamserver kamailio-3.3.2]# >> make all CC (gcc) [kamailio] receive.o >> LD (gcc) [kamailio] kamailio >> receive.o: In function `receive_msg': >> /root/kamailio-3.3.2/receive.c:163: undefined reference to >> `b2b_dlg_reply' >> collect2: ld returned 1 exit status >> make: *** [kamailio] Error 1 >> >> >> >> >> My doubt: >> Where should my b2bentities.so should be linked during compiling time? >> Please tell me the steps for it. >> >> >> Thanks & Regards, >> Manjusha A. >> Integra Micro Software Services (P) Ltd. >> >> >> >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> > seems that you want to create your custom module. have you checked the > makefile of your module. in order to develop the custom module just copy > the existing module and make relevant changes there. this is genuine way . > for this u need to include makefile.lst ,Makefile in your module. so best > way is to copy the smallest module (e.g.) pdb module and make changes. > > > > > ------------------------------ > > > Message: 3 > Date: Fri, 08 Feb 2013 09:27:07 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Multiple radius responses on a missed call > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114B6DB.10605(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Hello, > > > there are different event logged for accounting: - missed call event which > is sent from the point of view of callee - transaction answered which is > sent from the point of view of caller > > In db (eg mysql) accounting, each event in stored in a different table. > I don't know how they are in radius. You probably get two missed call > events and one transaction answered. > > If you want only one, do not set the flag for missed calls. > > > Cheers, > Daniel > > > On 2/7/13 9:23 PM, Efelin Novak wrote: > >> Hi folks, >> >> >> I'd like to ask you how can I solve an issue when I'm receiving three >> RADIUS requests on one missed call. My scenario is as follows: >> >> >> A calls to B. B returns 503 error message, INIVITE is then sent to C >> and C sends 486. >> >> After first 503 I receive a first RADIUS request. This is correct. >> >> >> However after 486 I receive three RADIUS requests with different IDs. >> As you can see in a DEBUG one RADIUS is send after the SIP reply >> message is received and second and third RADIUS messages are sent after >> the SIP reply message is forwarded. >> >> I'd like to receive only two RADIUS messages. One after 503 and second >> after 486. >> >> >> FLT_ACC is defined as 1. My config is as follows: >> modparam("acc", "failed_transaction_flag", FLT_ACC) modparam("acc", >> "report_cancels", 1) >> modparam("acc", "report_ack", 0) modparam("acc", >> "detect_direction", 0) >> modparam("acc", "early_media", 0) modparam("acc", "log_level", >> 5) >> modparam("acc", "log_flag", 1) modparam("acc", >> "log_missed_flag", 1) >> modparam("acc_radius", "radius_config", >> "/etc/radiusclient-ng/radiusclient.conf") # This is the >> location of the configuration file of radius client >> modparam("acc_radius", "radius_flag", FLT_ACC) >> modparam("acc_radius", "radius_missed_flag", FLT_ACC) >> >> In a route I set setflag("FLT_ACC") only once per call. >> >> >> Here is the debug for debug level 7: >> 0(5890) DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): >> 0(5890) DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> >> 0(5890) DEBUG: <core> [parser/msg_parser.c:638]: status: <486> >> 0(5890) DEBUG: <core> [parser/msg_parser.c:640]: reason: <Busy here> >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 232, <branch> = <z9hG4bK8d9c.4438d232.1>; state=6 >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 234, <received> = <192.168.21.101>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=2 0(5890) DEBUG: <core> [parser/msg_parser.c:513]: >> parse_headers: this >> is the first via 0(5890) DEBUG: <core> [receive.c:149]: After >> parse_msg... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: >> msg id=6 global id=5 T start=0xffffffff 0(5890) DEBUG: <core> >> [parser/parse_via.c:1286]: Found param type >> 232, <branch> = <z9hG4bK8d9c.e7b650e1.0>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=22 0(5890) DEBUG: <core> [parser/msg_parser.c:524]: >> parse_headers: this >> is the second via 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found >> param type 232, <branch> = <z9hG4bK463ae086>; state=6 >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 235, <rport> = <5060>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=22 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: >> add_param: tag=as493d6134 >> 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, >> state=29 0(5890) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: >> get_hdr_field: <To> [42]; uri=[sip:5556003900000@dr.vm] >> 0(5890) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body >> [<sip:5556003900000@dr.vm>] >> 0(5890) DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq >> <CSeq>: <102> <INVITE> >> 0(5890) DEBUG: tm [t_lookup.c:965]: DEBUG: t_reply_matching: hash >> 51672 label 590185284 branch 1 >> 0(5890) DEBUG: tm [t_lookup.c:1016]: DEBUG: t_reply_matching: reply >> matched (T=0x871afc98)! 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: >> trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: acc >> [acc_logic.c:537]: acc callback called for >> t(0x871afc98) event type 2, reply code 486 0(5890) DEBUG: tm >> [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 2, id 0 entered 0(5890) DEBUG: <core> [parser/parse_to.c:178]: >> DEBUG: add_param: tag=as5c1d222e >> 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, >> state=29 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... >> 0(5890) DEBUG: tm [t_lookup.c:1148]: DEBUG: t_check_msg: msg id=6 >> global id=6 T end=0x871afc98 0(5890) DEBUG: tm [t_reply.c:2081]: DEBUG: >> reply_received: org. >> status uas=180, uac[1]=180 local=0 is_invite=1) 0(5890) DEBUG: tm >> [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 4194304, id 0 entered 0(5890) DEBUG: siptrace [siptrace.c:794]: >> storing info... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: >> msg id=6 global id=6 T start=0x871afc98 0(5890) DEBUG: tm >> [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! >> 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> >> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [REPLY_TWO] Incoming reply code: 486 0(5890) DEBUG: tm >> [t_reply.c:1204]: ->>>>>>>>> T_code=180, new_code=486 >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 128, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback >> called for t(0x871afc98) event type 128, reply code 486 0(5890) acc >> [acc.c:275]: ACC: call missed: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius >> 0(5890) DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: >> 6e1b9f35a8b3e65859e29aefc1c26fc3 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=262 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_TYPE=application/sdp >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_SUPPORTED=replaces >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_ALLOW=INVITE, ACK, >> CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_DATE=Thu, 07 Feb 2013 >> 19:35:53 GMT >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_MAX_FORWARDS=68 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_USER_AGENT=Asterisk PBX >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=102 INVITE >> 0(5890) DEBUG: exec [exec_hf.c:267]: >> SIP_HF_CALLID=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> 0(5890) DEBUG: exec [exec_hf.c:267]: >> SIP_HF_CONTACT=<sip:0221028400@192.168.21.106> >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<sip:5556003900000@dr.vm> >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM="Janko" >> <sip:0221028400@dr.vm>;tag=as5c1d222e >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP >> 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0,SIP/2.0/UDP >> 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 >> 0(5890) DEBUG: exec [exec_mod.c:175]: executing >> [/opt/kamrouter/scripts/notify_reroute.sh 503 >> sip:3333003900000@ast-car.vm sip:0221028400@dr.vm >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm] >> sh: /opt/kamrouter/scripts/notify_reroute.sh: not found >> 0(5890) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on >> 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=0 >> global id=0 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: >> DEBUG: t_check_msg: T already found! >> 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> >> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [FAIL_TWO] Rerouted gw returned 486 0(5890) INFO: <script>: XLOG: >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [aaa] VYMAZAVAM3 0(5890) DEBUG: >> tm [t_reply.c:1702]: DEBUG: relay_reply: branch=1, save=0, relay=1 >> 0(5890) DEBUG: <core> [msg_translator.c:1933]: old size: 572, new >> size: 485 >> 0(5890) DEBUG: <core> [msg_translator.c:1951]: >> build_res_from_sip_res: copied size: orig:110, new: 23, rest: 462 msg= >> SIP/2.0 486 Busy here >> Via: SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0 >> Via: SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 >> From: "Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e >> To: <sip:5556003900000@dr.vm>;tag=as493d6134 >> Call-ID: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> CSeq: 102 INVITE >> Server: Asterisk PBX 11.0.2 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO, PUBLISH >> Supported: replaces, timer >> Content-Length: 0 >> >> >> >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 1048576, id 0 entered 0(5890) DEBUG: dialog [dlg_hash.c:588]: ref >> dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:590]: >> dialog id=3894 found on entry 2825 0(5890) DEBUG: dialog >> [dlg_hash.c:940]: dialog 0x871af3f0 changed >> from state 2 to state 5, due event 4 (ref 2) 0(5890) DEBUG: dialog >> [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 3 >> 0(5890) DEBUG: dialog [dlg_handlers.c:1469]: executing event_route 4 on >> state 5 0(5890) INFO: <script>: XLOG: >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [dialog:failed] Deleting 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm from >> dialogs and dstprefix_limit 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref >> dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_handlers.c:543]: >> dialog 0x871af3f0 failed (negative reply) >> 0(5890) DEBUG: dialog [dlg_cb.c:256]: dialog=0x871af3f0, type=4 >> 0(5890) DEBUG: dialog [dlg_hash.c:643]: ref dlg 0x871af3f0 with 1 -> 3 >> 0(5890) DEBUG: dialog [dlg_hash.c:646]: dialog >> callid='38e422e10af08b2e34d58e5b765e81d3(a)dr.vm' found on entry 2825, >> dir=1 0(5890) DEBUG: dialog [dlg_var.c:203]: Internal var-list ((nil)): >> 0(5890) DEBUG: dialog [dlg_var.c:213]: Dialog var-list (0x871af6f8): >> 0(5890) DEBUG: dialog [dlg_var.c:218]: "ru"=sip:003900000@cr1.vm:5060 >> (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: d=0 (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: et=1360265682.889 (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: st=1360265682.889 (flags 1) >> 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 >> 0(5890) acc [acc_cdr.c:192]: st=1360265682.889; et=1360265682.889; >> d=0; c1= 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 >> with 1 -> 1 0(5890) DEBUG: dialog [dlg_handlers.c:937]: registering TMCB >> to wait for negative ACK 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg >> 0x871af3f0 with 1 -> 2 >> 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 512, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback >> called for t(0x871afc98) event type 512, reply code 486 0(5890) acc >> [acc.c:275]: ACC: call missed: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius >> 0(5890) acc [acc.c:275]: ACC: transaction answered: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:867]: acc event for engine: radius >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 8388608, id 0 entered >> >> Thanks for a help. >> >> >> Efelin >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > > ------------------------------ > > > Message: 4 > Date: Fri, 08 Feb 2013 09:36:01 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Reg. shared library linking during compiling > time To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114B8F1.8020807(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Hello, > > > some guidelines to write a new module are available at: - > http://www.asipto.com/pub/kamailio-devel-guide/#c16_devel_new_module > > > Cheers, > Daniel > > > On 2/8/13 8:06 AM, kiran bhosale wrote: > >> madhumanjusha(a)integramicro.com wrote: >>> Hello All, >>> Please find below steps...and kindly let me know the error solution. >>> >>> >>> STEP 1:: >>> Under modules I created a directory name b2bentities >>> /root/kamailio-3.3.2/modules/b2bentities >>> $ls >>> b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o >>> dialog.h librpath.lst makecfg.lst Makefile >>> >>> In b2b_server.h >>> #ifndef SERVER_H >>> #define SERVER_H >>> #endif >>> >>> >>> int b2b_dlg_reply(); >>> >>> >>> >>> In b2b_server.c I wrote following code :: >>> >>> >>> int b2b_dlg_reply() { >>> LOG(L_INFO,"Hello am in b2bua module\n"); >>> return 0; } >>> >>> >>> Also,I added below code to the same file as per steps given in >>> kamailio server's developer's guide. In developer's guide,they told to >>> initialize parameters as per module's requirement but i made them all >>> empty and null... for example static int mod_init(void) { >>> LOG(L_INFO,"IN mod_init\n"); >>> return 0; } >>> >>> >>> static cmd_export_t commands[] = { }; >>> >>> >>> >>> static param_export_t params[]={ }; >>> >>> >>> struct module_exports exports = { "b2bentities", // module name >>> NULL, // dlopen flags >>> commands, // exported functions params, // exported >>> parameters NULL, // exported statistics >>> NULL, // exported MI functions >>> NULL, // exported pseudo-variables >>> NULL, // extra processes >>> mod_init, // module init function (before fork. kids will >>> inherit) NULL, // reply processing function >>> NULL, // destroy function >>> child_init // child init function }; >>> >>> >>> >>> STEP 2:: >>> In kamailio core >>> file : In receive.c Method Name : In receive_msg >>> >>> >>> i did this way... >>> >>> In receive.c add follow piece of code:: >>> >>> >>> #include "modules/b2bentities/b2b_server.h" >>> >>> >>> int receive_msg(....) { >>> /*my code*/ >>> int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; >>> if(msg->REQ_METHOD!=METHOD_REGISTER) { >>> >>> >>> b2b_ret=(*ptr)(); if(b2b_ret!=0) { >>> LOG(L_INFO,"B2B Dialogue reply function >>> failed\n"); exit(1); } >>> else { >>> LOG(L_INFO,"B2B Dialogue reply function >>> success\n"); exit(1); } >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> STEP 3:: >>> I even added kamailio.cfg load module "b2bentities.so" >>> >>> >>> >>> >>> STEP 4:: >>> Final step,when I do make all..... >>> its giving me the following error :: [root@kamserver kamailio-3.3.2]# >>> make all CC (gcc) [kamailio] receive.o >>> LD (gcc) [kamailio] kamailio >>> receive.o: In function `receive_msg': >>> /root/kamailio-3.3.2/receive.c:163: undefined reference to >>> `b2b_dlg_reply' >>> collect2: ld returned 1 exit status >>> make: *** [kamailio] Error 1 >>> >>> >>> >>> >>> My doubt: >>> Where should my b2bentities.so should be linked during compiling time? >>> Please tell me the steps for it. >>> >>> >>> Thanks & Regards, >>> Manjusha A. >>> Integra Micro Software Services (P) Ltd. >>> >>> >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>> list sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> seems that you want to create your custom module. have you checked the >> makefile of your module. in order to develop the custom module just copy >> the existing module and make relevant changes there. this is genuine way >> . for this u need to include makefile.lst ,Makefile in >> your module. so best way is to copy the smallest module (e.g.) pdb module >> and make changes. >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > > ------------------------------ > > > _______________________________________________ > sr-users mailing list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > End of sr-users Digest, Vol 93, Issue 13 > **************************************** > > Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

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Re: [SR-Users] [sr-dev] openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Hi Shahzad , It is not ,We can able to connect remote mysql ( please find the detail below),moreover we can able create openser db from Kamailio server to remote mysql server *Mysql server : 192.168.1.170* *Kamailio server : 192.168.1.101* *Mysql output :* * * *mysql -u openser -h 192.168.1.170 -p* Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 68 Server version: 5.1.67 Source distribution Copyright (c) 2000, 2012, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql> With Regards N.Prakash > > ---------- Forwarded message ---------- > From: Muhammad Shahzad <shaheryarkh(a)gmail.com> > Date: Tue, Feb 12, 2013 at 5:16 PM > Subject: Re: [sr-dev] [SR-Users] openser db on an external mysql server > To: Development mailing list of the sip-router project < > sr-dev(a)lists.sip-router.org> > Cc: Daniel-Constantin Mierla <miconda(a)gmail.com>, > sr-users(a)lists.sip-router.org, Indu Priya <indu(a)tevatel.com>, Prabhakaran > B <prabhakaran(a)tevatel.com> > > > Its not a Kamailio error, it is problem with connecting to mysql server. > Error 4 means No MySQL Server Running On Specified Host. > > May be MySQL server is down or there is some firewall running or MySQL not > listening on IP you specified. > > Thank you. > > > On Tue, Feb 12, 2013 at 12:11 PM, Prakash N <prakash.n(a)tevatel.com> wrote: > >> Dear Daniel, >> >> Thanks for your quick response >> >> We can able to create operseb db on remote mysql server ,but >> Kamailio not getting start >> >> Error detail: >> >> ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL >> server on '192.168.1.170' (4) >> ERROR: <core> [db.c:32 0]: could not add connection to the pool >> ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database >> ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while >> initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) >> >> >> I have attached my kamailo.cfg ( we followed Kamailio and Asterisk real >> time installation) ,please advice >> >> With Regards >> >> N.Prakash >> >>> >>> ---------- Forwarded message ---------- >>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>> Date: Tue, Feb 12, 2013 at 2:41 PM >>> Subject: Re: [SR-Users] openser db on an external mysql server >>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>> Users Mailing List" <sr-users(a)lists.sip-router.org>, >>> sr-dev(a)lists.sip-router.org, miconda(a)gmail.com >>> Cc: Prabhakaran B <prabhakaran(a)tevatel.com> >>> >>> >>> Hello, >>> >>> kamailio database is created with: >>> >>> kamdbctl create >>> >>> the host of mysql server has to be specified in kamctlrc file via DBHOST >>> variable - kamctlrc is located in same directory as kamailio.cfg. >>> >>> In kamailio.cfg, update the db_url value (or DBURL define) to reflect >>> the mysql server host. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 2/12/13 4:16 AM, Prakash N wrote: >>> >>> Hi All, >>> >>> I am new this Kamailo ,ther is any step by step document to set >>> up openser db on an external mysql server ? >>> >>> With Regards >>> >>> N.Prakash >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio World Conference, April 16-17, 2013, Berlin >>> - http://conference.kamailio.com - >>> >>> >>> >> >> _______________________________________________ >> sr-dev mailing list >> sr-dev(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk(a)hotmail.com > Email: shaheryarkh(a)googlemail.com > >

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Re: [SR-Users] openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Dear Daniel, Thanks for your quick response We can able to create operseb db on remote mysql server ,but Kamailio not getting start Error detail: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) ERROR: <core> [db.c:32 0]: could not add connection to the pool ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) I have attached my kamailo.cfg ( we followed Kamailio and Asterisk real time installation) ,please advice With Regards N.Prakash > > ---------- Forwarded message ---------- > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Date: Tue, Feb 12, 2013 at 2:41 PM > Subject: Re: [SR-Users] openser db on an external mysql server > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users > Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev(a)lists.sip-router.org, > miconda(a)gmail.com > Cc: Prabhakaran B <prabhakaran(a)tevatel.com> > > > Hello, > > kamailio database is created with: > > kamdbctl create > > the host of mysql server has to be specified in kamctlrc file via DBHOST > variable - kamctlrc is located in same directory as kamailio.cfg. > > In kamailio.cfg, update the db_url value (or DBURL define) to reflect the > mysql server host. > > Cheers, > Daniel > > > On 2/12/13 4:16 AM, Prakash N wrote: > > Hi All, > > I am new this Kamailo ,ther is any step by step document to set > up openser db on an external mysql server ? > > With Regards > > N.Prakash > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > >

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Fwd: INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers
by Bayram Karagoz 12 Feb '13

12 Feb '13

---------- Forwarded message ---------- From: Bayram Karagoz <karagoz.bayram(a)gmail.com> Date: 2013/2/12 Subject: Re: [SR-Users] INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers To: miconda(a)gmail.com Cc: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> recheck config and saw that I used dlg_manage() most before route(QUOTA) block. does it sits in the wrong place? you can find my kamailio.cfg file in attachment also. # account only INVITEs * dlg_manage();* if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } ###Display Name manipulation### if (is_method("INVITE") && ($fn != $fU)) { uac_replace_from("$fU",""); } ### incoming call from SBC### route(INCOMINGSBC); ###Suspend User### route(SUSPEND); ### Concurrent Call Limitation Routing Block ### * route(QUOTA);* 2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com> > Where do you use dlg_manage()? It is not in the route you pasted below. > > Cheers, > Daniel > > > On 2/12/13 10:12 AM, Bayram Karagoz wrote: > > Thanks for quick response Daniel, > > My kamailio version is 3.2.3 (x86_64/linux) . > I want to use dlg_manage() before but not sure which side of my call > limitation route code block. > can you advise me where to use dlg_manage() in my routing? > here is my route block defined below; > > > route[QUOTA] { > > if (is_method("INVITE") && $fU == $avp(s:username)) { > > avp_db_query("select quota from subscriber where > username='$avp(s:username)'", "$avp(s:quota)"); > > $var(SIZE) = 0; > get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); > xlog("L_INFO", "CALL_KAMAILIO:Call originated from = > $fd\n"); > xlog("L_INFO", "Active call amount for $fU is $var(SIZE) , > call-id=$ci, quota=$avp(s:quota)\n"); > if( $var(SIZE) >= $avp(s:quota) ){ > sl_send_reply("403", "Simultaneous calls limit reached"); > xlog("L_INFO", "QUOTA:Limit of $var(SIZE) active calls has > been reached for user=$avp(s:username), call-id=$ci, > quota=$avp(s:quota)\n"); > exit; > } > set_dlg_profile("callquota","$avp(s:username)"); > } > } > > > 2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com> > >> Hello, >> >> what version of kamailio are you using? >> >> State 1 is for dialogs that were not relayed (or they haven't received >> any reply yet). That means you create the dialog and don't relay the >> INVITE. Try to us dlg_manage() just before relaying the INVITE. >> >> Cheers, >> Daniel >> >> >> On 2/12/13 9:37 AM, Bayram Karagoz wrote: >> >> Hi, >> >> is there any advise about my problem? >> I'm using kamailio in my production environment and this issue started to >> reproduce a lot. >> is there any way to discard retransmitted INVITE messages from counting >> dialog? >> >> >> 2013/2/9 Bayram Karagoz <karagoz.bayram(a)gmail.com> >> >>> thanks for the response Daniel, >>> >>> has_totag not solved my problem. >>> >>> in a deep search I realized that these retransmission INVITE dialogs >>> are waiting in dialog list as state = 1. one example is below; >>> is there any way to clear these dialog which state is waiting for state >>> = 1 or any other solution is welcome for me? >>> >>> dialog:: hash=246:499719317 >>> state:: 1 >>> ref_count:: 1 >>> timestart:: 0 >>> timeout:: 0 >>> callid:: 1356619-3569384775-474862(a)GBSBC-1.mydomain.com >>> from_uri:: sip:02165151057@xxx.xxx.xxx.xxx >>> from_tag:: 3569384775-474868 >>> caller_contact:: sip:02165151057@xxx.xxx.xxx.xxx:5060 >>> caller_cseq:: 2 >>> caller_route_set:: >>> caller_bind_addr:: udp:xxx.xxx.xxx.xxx:5060 >>> callee_bind_addr:: >>> to_uri:: sip:4115667@xxx.xxx.xxx.xxx >>> to_tag:: >>> callee_contact:: >>> callee_cseq:: >>> callee_route_set:: >>> >>> >>> >>> >>> 2013/2/4 Daniel Tryba <daniel(a)pocos.nl> >>> >>>> On Sunday 03 February 2013 12:58:02 Bayram Karagoz wrote: >>>> > I'm trying to limit all subscriber's active call according to setting >>>> their >>>> > simultaneous use setting defined in database. >>>> > But having problem when one subscriber's INVITE message goes to >>>> > retransmission. When retranmission occures INVITE messages don't >>>> complete >>>> > with BYE and subscriber simultaneus use amount will hung. >>>> > My example code block is below. >>>> > Does anyone advice me for overcome INVITE retransmissions? >>>> >>>> I haven't implemented it myself yet, but all examples I've seen check >>>> for >>>> !has_totag() in the INVITE when doing call limit accounting. So >>>> if (is_method("INVITE") && $fU == $avp(s:username) && !has_totag()) >>>> in your case. But that only works if the device received the initial 100 >>>> Trying with a tag from Kamailio. >>>> >>>> But might it not be better to just do the count in the INVITE and >>>> adding/deleting of dialogs in the dialog event routes? >>>> http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2556156 >>>> Problem here is that if the limit is 5 and and there are more than 5 >>>> concurrent INVITES you'd have to drop calls with a 200 OK response. >>>> >>>> -- >>>> >>>> POCOS B.V. - Croy 9c - 5653 LC Eindhoven >>>> Telefoon: 040 293 8661 - Fax: 040 293 8658 >>>> http://www.pocos.nl/ - http://www.sipo.nl/ >>>> K.v.K. Eindhoven 17097024 >>>> >>> >>> >>> >>> -- >>> *Bayram KARAGÖZ* >>> >>> *NGN Engineer* >>> >>> *-----------------------------------------* >>> >>> *@* bayram(a)bayramkaragoz.org >>> >>> *:** *www.bayramkaragoz.org >>> >>> *-----------------------------------------* >>> >> >> >> >> -- >> *Bayram KARAGÖZ* >> >> *NGN Engineer* >> >> *-----------------------------------------* >> >> *@* bayram(a)bayramkaragoz.org >> >> *:** *www.bayramkaragoz.org >> >> *-----------------------------------------* >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio World Conference, April 16-17, 2013, Berlin >> - http://conference.kamailio.com - >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > -- > *Bayram KARAGÖZ* > > *NGN Engineer* > > *-----------------------------------------* > > *@* bayram(a)bayramkaragoz.org > > *:** *www.bayramkaragoz.org > > *-----------------------------------------* > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > -- *Bayram KARAGÖZ* *NGN Engineer* *-----------------------------------------* *@* bayram(a)bayramkaragoz.org *:** *www.bayramkaragoz.org *-----------------------------------------* -- *Bayram KARAGÖZ* *NGN Engineer* *-----------------------------------------* *@* bayram(a)bayramkaragoz.org *:** *www.bayramkaragoz.org *-----------------------------------------*

2 2

Kamailio World Updates
by Daniel-Constantin Mierla 12 Feb '13

12 Feb '13

Hello, we have published the first set of accepted speeches at Kamailio World Conference, you can view the current draft of the schedule at: - http://conference.kamailio.com/k01/schedule/ There are already about 15 speakers and presentations in the agenda for the two days of conference, other proposals being in the review queue. Another good news is that business environment around Kamailio project is growing, the sponsors of Kamailio World Conference, Sipwise, Sipgate and Asipto are looking to hire or subcontract new talents, check the Career & Jobs page at: - http://conference.kamailio.com/k01/career-jobs/ Greetings again to our community members that participated at Fosdem, we were quite a big group there, unfortunately not much time to catch up in detail with everyone. Looking forward to see many of you in Berlin as well, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Handling 302 redirect on group ringing
by Mark Boyce 12 Feb '13

12 Feb '13

Hi All Got a problem handling redirects when more than one phone is ringing. The setup is this; Phone1 - Extension 100 Phone 2 - Extension 101 Phone 3 - Extension 102 & 101 When Phone 1 rings Extension 101 Phones 2 & 3 ring correctly as you would expect. If Phone 2 now sets the on phone forward/redirect we see the following; Phone 1 - Invite sent to 101 Phone 2 & 3 Receive Invites Phone 2 responds with a 302 Redirect Phone 3 Rings. Which again is as you would expect. The problem is that the 302 Message we're getting back is being seen in onreply_route where I cannot add a new branch to handle it. (Error - misused command append_branch / Command cannot be used in the block) If Phone 3 is unavailable / times out / busy / or otherwise does not take the call we then see the 302 from Phone 2 arrive in failure_route where it is handled as expected. The call flow we're trying to achieve is that the redirect is followed whilst the other available phone carries on ringing. I've tried with failure_reply_mode as the default 0 and as 3, both do the same thing. Currently working with kamailio version 3.4.0-dev6 Can anyone spot what I'm doing wrong ? Thanks Mark

2 1

How to detect there are no matching subscriber for uri
by Krishna Kurapati 12 Feb '13

12 Feb '13

Hi, When we receive an request, we do " lookup("location") call to find if the uri is there or not. There are two scenarios: 1. The aor is in the subscriber table but not in usrloc table 2. The aor is not in subscriber table I would like to respond differently to the sender in each of these cases. Right now, the lookup returns -1 for either case. Is there any other way other than dipping into database each time? Thanks Krish Kura

2 1

INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers
by Bayram Karagoz 12 Feb '13

12 Feb '13

Hello, I'm trying to limit all subscriber's active call according to setting their simultaneous use setting defined in database. But having problem when one subscriber's INVITE message goes to retransmission. When retranmission occures INVITE messages don't complete with BYE and subscriber simultaneus use amount will hung. My example code block is below. Does anyone advice me for overcome INVITE retransmissions? Thanks. modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) modparam("dialog","profiles_with_value","callquota") modparam("dialog", "default_timeout", 3600) modparam("dialog", "dlg_match_mode",0) modparam("dialog", "detect_spirals", 1) route[QUOTA] { if (is_method("INVITE") && $fU == $avp(s:username)) { avp_db_query("select quota from subscriber where username='$avp(s:username)'", "$avp(s:quota)"); $var(SIZE) = 0; get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); xlog("L_INFO", "CALL_OPENSER:Call originated from = $fd\n"); xlog("L_INFO", "Active call amount for $fU is $var(SIZE) , quota=$avp(s:quota)\n"); if( $var(SIZE) >= $avp(s:quota) ){ sl_send_reply("403", "Simultaneous calls limit reached"); xlog("L_INFO", "QUOTA:Limit of $var(SIZE) active calls has been reached for user=$avp(s:username), quota=$avp(s:quota)\n"); exit; } set_dlg_profile("callquota","$avp(s:username)"); get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); } } -- *Bayram KARAGÖZ*

3 6

Video Conferencing
by Mohakem Khan 12 Feb '13

12 Feb '13

Does kamailio support video endpoint/codecs? How about video conference? Is there MCD support?

3 2

openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Hi All, I am new this Kamailo ,ther is any step by step document to set up openser db on an external mysql server ? With Regards N.Prakash

2 1

mysql: driver_error
by Uri Shacked 11 Feb '13

11 Feb '13

Hi, I am doing some testing and i have trouble understanding when does the "mysql:driver_error" counter/statistics increase.... I stopped the mysqld process on the DB server and used sqlops. the driver_error did still is on 0. I use version 3.3.2. Thanks, Uri

2 3

Re: [SR-Users] dlg_manage and engage_media_proxy
by Daniel-Constantin Mierla 11 Feb '13

11 Feb '13

Hello, On 2/11/13 5:17 PM, Eduardo Lejarreta wrote: > Good evening Daniel. > > You are right, calling engage_media_proxy() prior to dlg_manage() solves the > issue (just before t_relay on INVITEs). > > I've done several test changing the order of things and now everything works > fine. great, thanks for reporting back. Perhaps dlg flag vs dlg_manage has to be clarified a bit more in the next major release (4.1). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio multiple registration issue
by Ramaseshi reddy kolli 11 Feb '13

11 Feb '13

Hi, i am using kamailio 3.3 version please help me with this issue. 1) Registered from SIP end point, kamailio added row in location table 2) Now using same SIP account details, pushed registration entry using *kamctl ul add , * instead of overwriting the previous contact by new one it inserts another new row in the location table 3) Total now 2 rows with same username are present in DB. I am using save("location", "0x04") so before inserting new row it should check whether already entry is there or not, if there should update the same row. I look forward to your reply. Thanks Ram

2 2

Re: [SR-Users] dlg_manage and engage_media_proxy
by Daniel-Constantin Mierla 11 Feb '13

11 Feb '13

On 2/8/13 12:15 PM, Eduardo Lejarreta wrote: > Yes it's the correct order. We call dlg_manage() prior to > engage_media_proxy(): The correct order could be the other one, as mediaproxy might need some callbacks from dialog module. Usually the dlg_manage() should be executed just before relaying, when is no other chance that the invite will get a negative reply from the config. Cheers, Daniel > > route[INVITE_MANAGE] { > > if (is_method("INVITE")) { > > $dlg_var(si) = $si; > $dlg_var(tU) = $tU; > $dlg_ctx(timeout_bye) = 1; > > dlg_manage(); > ........ > ........ > } > } > ....... > ....... > route[RELAY] { > > if (is_method("INVITE")) { > ........ > ........ > route(MEDIAPROXY); # route MEDIAPROXY > t_on_failure(FAIL_ONE); > t_on_reply(REPLY_ONE); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > ........ > ........ > route[MEDIAPROXY] { > > xlog("L_INFO", "Engage Media Proxy - M=$rm R=$ru F=$fu T=$tu > IP=$si:$sp ID=$ci\n"); > engage_media_proxy(); > return; > } > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

UC Expo London
by Peter Dunkley 11 Feb '13

11 Feb '13

Hello, Are there any plans for a Kamailio meet-up on one of the evenings this year? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Kamailio-Java Integration
by ron.kamailio＠mcleodnet.com 10 Feb '13

10 Feb '13

I am looking for a way to integrate Java business logic with Kamailio to influence routing decisions, add/remove/modify headers, record accounting-related information, etc. A while back there was a project named WeSIP, but it doesn't appear to active anymore. Is anyone doing something like this now, or any ideas or suggestions? Thanks, Ron

8 23

Multiple radius responses on a missed call
by Efelin Novak 08 Feb '13

08 Feb '13

Hi folks, I'd like to ask you how can I solve an issue when I'm receiving three RADIUS requests on one missed call. My scenario is as follows: A calls to B. B returns 503 error message, INIVITE is then sent to C and C sends 486. After first 503 I receive a first RADIUS request. This is correct. However after 486 I receive three RADIUS requests with different IDs. As you can see in a DEBUG one RADIUS is send after the SIP reply message is received and second and third RADIUS messages are sent after the SIP reply message is forwarded. I'd like to receive only two RADIUS messages. One after 503 and second after 486. FLT_ACC is defined as 1. My config is as follows: modparam("acc", "failed_transaction_flag", FLT_ACC) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) modparam("acc", "detect_direction", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 5) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # This is the location of the configuration file of radius client modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACC) In a route I set setflag("FLT_ACC") only once per call. Here is the debug for debug level 7: 0(5890) DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): 0(5890) DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> 0(5890) DEBUG: <core> [parser/msg_parser.c:638]: status: <486> 0(5890) DEBUG: <core> [parser/msg_parser.c:640]: reason: <Busy here> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK8d9c.4438d232.1>; state=6 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 234, <received> = <192.168.21.101>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 0(5890) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via 0(5890) DEBUG: <core> [receive.c:149]: After parse_msg... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6 global id=5 T start=0xffffffff 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK8d9c.e7b650e1.0>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=22 0(5890) DEBUG: <core> [parser/msg_parser.c:524]: parse_headers: this is the second via 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK463ae086>; state=6 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 235, <rport> = <5060>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=22 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=as493d6134 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 0(5890) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_field: <To> [42]; uri=[sip:5556003900000@dr.vm] 0(5890) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body [<sip:5556003900000@dr.vm>] 0(5890) DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq <CSeq>: <102> <INVITE> 0(5890) DEBUG: tm [t_lookup.c:965]: DEBUG: t_reply_matching: hash 51672 label 590185284 branch 1 0(5890) DEBUG: tm [t_lookup.c:1016]: DEBUG: t_reply_matching: reply matched (T=0x871afc98)! 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 2, reply code 486 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=as5c1d222e 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... 0(5890) DEBUG: tm [t_lookup.c:1148]: DEBUG: t_check_msg: msg id=6 global id=6 T end=0x871afc98 0(5890) DEBUG: tm [t_reply.c:2081]: DEBUG: reply_received: org. status uas=180, uac[1]=180 local=0 is_invite=1) 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 4194304, id 0 entered 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6 global id=6 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [REPLY_TWO] Incoming reply code: 486 0(5890) DEBUG: tm [t_reply.c:1204]: ->>>>>>>>> T_code=180, new_code=486 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 128, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 128, reply code 486 0(5890) acc [acc.c:275]: ACC: call missed: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius 0(5890) DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: 6e1b9f35a8b3e65859e29aefc1c26fc3 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=262 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_TYPE=application/sdp 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_SUPPORTED=replaces 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_ALLOW=INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_DATE=Thu, 07 Feb 2013 19:35:53 GMT 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_MAX_FORWARDS=68 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_USER_AGENT=Asterisk PBX 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=102 INVITE 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CALLID=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTACT=<sip:0221028400@192.168.21.106> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<sip:5556003900000@dr.vm> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM="Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0,SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 0(5890) DEBUG: exec [exec_mod.c:175]: executing [/opt/kamrouter/scripts/notify_reroute.sh 503 sip:3333003900000@ast-car.vm sip:0221028400@dr.vm 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm] sh: /opt/kamrouter/scripts/notify_reroute.sh: not found 0(5890) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=0 global id=0 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [FAIL_TWO] Rerouted gw returned 486 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [aaa] VYMAZAVAM3 0(5890) DEBUG: tm [t_reply.c:1702]: DEBUG: relay_reply: branch=1, save=0, relay=1 0(5890) DEBUG: <core> [msg_translator.c:1933]: old size: 572, new size: 485 0(5890) DEBUG: <core> [msg_translator.c:1951]: build_res_from_sip_res: copied size: orig:110, new: 23, rest: 462 msg= SIP/2.0 486 Busy here Via: SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0 Via: SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 From: "Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e To: <sip:5556003900000@dr.vm>;tag=as493d6134 Call-ID: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm CSeq: 102 INVITE Server: Asterisk PBX 11.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 1048576, id 0 entered 0(5890) DEBUG: dialog [dlg_hash.c:588]: ref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:590]: dialog id=3894 found on entry 2825 0(5890) DEBUG: dialog [dlg_hash.c:940]: dialog 0x871af3f0 changed from state 2 to state 5, due event 4 (ref 2) 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 3 0(5890) DEBUG: dialog [dlg_handlers.c:1469]: executing event_route 4 on state 5 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [dialog:failed] Deleting 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm from dialogs and dstprefix_limit 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_handlers.c:543]: dialog 0x871af3f0 failed (negative reply) 0(5890) DEBUG: dialog [dlg_cb.c:256]: dialog=0x871af3f0, type=4 0(5890) DEBUG: dialog [dlg_hash.c:643]: ref dlg 0x871af3f0 with 1 -> 3 0(5890) DEBUG: dialog [dlg_hash.c:646]: dialog callid='38e422e10af08b2e34d58e5b765e81d3(a)dr.vm' found on entry 2825, dir=1 0(5890) DEBUG: dialog [dlg_var.c:203]: Internal var-list ((nil)): 0(5890) DEBUG: dialog [dlg_var.c:213]: Dialog var-list (0x871af6f8): 0(5890) DEBUG: dialog [dlg_var.c:218]: "ru"=sip:003900000@cr1.vm:5060 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: d=0 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: et=1360265682.889 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: st=1360265682.889 (flags 1) 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 0(5890) acc [acc_cdr.c:192]: st=1360265682.889; et=1360265682.889; d=0; c1= 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 0(5890) DEBUG: dialog [dlg_handlers.c:937]: registering TMCB to wait for negative ACK 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 512, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 512, reply code 486 0(5890) acc [acc.c:275]: ACC: call missed: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius 0(5890) acc [acc.c:275]: ACC: transaction answered: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:867]: acc event for engine: radius 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 8388608, id 0 entered Thanks for a help. Efelin

2 4

Reg. functions in custom modules
by madhumanjusha＠integramicro.com 08 Feb '13

08 Feb '13

Hello All, Is it possible to call functions in custom modules say modules/my_module/my_file.c if my_file.c has a function and want to access/call it from kamailio core file ie., which is under /root/kamailio-3.3.2/receive.c ?? If not possible,what is the way to access functions in custom modules. Please do reply. Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 2

dlg_manage and engage_media_proxy
by Eduardo Lejarreta Langarica SARENET 08 Feb '13

08 Feb '13

Good evening. Kamailio 3.2.2 and mediaproxy 2.5.2 We find a problem when using engage_media_proxy and dlg_manage on the same kamailio.cfg (we loose one way audio) is ther any issue with regard to this? may be anything we're doing wrong? any other easy way to control concurrent calls? If 1.- we avoid dlg_manage with engage_media_proxy or 2.- we use dlg_manage with "use_media_proxy/end_media_session" every thing works fine. Any idea will be helpful. Thanks and best regards. ----------------------------------------------------------------------------------------------------------------------------------- Eduardo Lejarreta

3 2

kamctl restart and mediaproxy
by Eduardo Lejarreta Langarica SARENET 08 Feb '13

08 Feb '13

Good evening again. Kamailio 3.2.2 and mediaproxy 2.5.2 An long-time issue for us: kamctl restart stops mediaproxy audio. is there anyway to avoid this? Any idea will be helpful. Thanks and best regards. ----------------------------------------------------------------------------------------------------------------------------------- Eduardo Lejarreta

3 2

Somewhat OT: Specific SIP packets can cause ethernet controller reset
by Kristian Kielhofner 08 Feb '13

08 Feb '13

While not strictly Kamailio related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html I'm hearing initial reports that this issue is fairly widespread. -- Kristian Kielhofner

2 1

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Modify IP address in SDP (Austin Einter)
by madhumanjusha＠integramicro.com 08 Feb '13

08 Feb '13

On Tue, January 22, 2013 1:58 pm, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Tue, 22 Jan 2013 18:37:13 +0530 > From: Austin Einter <austin.einter(a)gmail.com> > Subject: [SR-Users] Modify IP address in SDP > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANXt1k--VtC6aYHGTBZ5GqF8n=66g5PVwG4YhOC_uefxpVYy=g(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Dear All > When 200 OK to INVITE passes through Kamailio proxy , I want to change the > IP address in SDP to a specific IP address. > > > Can this be done by just modifying configuration files or do I need to > modify in source code. > > In case I need to modify source code, which files I need to look at. > > > Kindly provide pointers for said problem. > > > Thanks > Austin Hello Austin, 1...I was doing similar one for my task. Files : /root/kamailio-3.3.2/parser/sdp/sdp.h sdp.h file has struct that holds ip address and port numbers of audio and video. Stucture: sdp_session_cell struct has member named str ip_addr will give you ip address in sdp. 2...Yes you can edit existing code or else you can create your own 200ok packet.I created my own 200OK and invite packets with sdp. 3...You can grab below ip address when kamailio proxy is about to forward 200OK to end client.File where you can find 200OK is (though not sure but you can try this file edit existing 200OK): /root/kamailio-3.3.2/modules/tm/t_reply.c has below function static int _reply( struct cell *trans, struct sip_msg* p_msg, unsigned int code, char * text, int lock ) { } 4...I did the following way to get ipaddr...and below piece of code you can use it in the above function's body to edit 200OK's ip address:: char my_src_rtp_addr[8]; int my_src_rtp_addr_len=0; struct sdp_session_cell *sdp_session; if(0 != parse_sdp(p_msg)) { LOG(L_INFO,"Unable to parse sdp\n"); } sdp_session=get_sdp_session(p_msg,0); if(sdp_session!=0){ my_src_rtp_addr_len=strlen(my_src_rtp_addr); my_src_rtp_addr[my_src_rtp_addr_len]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+1]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+2]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+3]='\0'; strncpy(my_src_rtp_addr,sdp_session->ip_addr.s,sdp_session->ip_addr.len); } If any questions,please let me know. Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

3 3

Need help in rewriting the Request URI
by Ajay Sabat 07 Feb '13

07 Feb '13

Hi, I use Kamailio for basic call setup with one exception where it has to route call to another server by seeing the domain name in the Request URI. If the Request uri in the received INVITE message looks like "sip12345@video-conf", then it should route the INVITE request to the video-conf server by resolving the domain name. I changed the config file kamailio.cfg as given below. route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); # AKS: call routing if (uri=~".*@video-conf") { rewritehost("video-conf"); } # AKS route(RELAY); } } It resolves the domain name and routes the INVITE message to the resolved IP address. However it uses the domain name in the Request line and To header and I need to replace domain with the resolved IP address in the Request URI and To header. Now Request URI and To header look like "INVITE sip:12345@video-conf" and "To: <sip:12345:video-conf>" respectively and I need them as "INVITE sip:12345@video-conf" and "To: <sip:12345:video-conf>". I don't have any knowledge in tweaking kamailio. I tried some tricks like the following by going through documents and pervious emails in the forum. But it didn't work. Some of them have spoken about avp and pseudo variables. But I am getting confused as I have just begun using Kamailio and it is vast. # $ru = "sip:" + $rU + "@" + to_ip; I will appreciate if you could help me or guide me in right direction. Thanks, Ajay

2 2

Problems with MSILO module can't receive messages with m_dump.
by Nord7 07 Feb '13

07 Feb '13

Hello i have two problems with module MSILO and text messages from SIP client BRIA. 1. I can't send messages with m_dump. I have installed Kamailio 3.3 in default configuration. I'm added "m_dump("$fu")" to my config in route[REGISTRAR]. If i use client CSipSimple for sending messages to offline user "333". i'm receive notification message "*** User 333 offline!". Next i connect to Kamailio with user "333" and i see in kamailio log what "m_dumt" trying to send messages but can't because "request authorisation". How to disable authorisation request, without leak security. route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); if (m_dump("$fu")) { log("MSILO: offline messages dumped - if they were\n"); } else { log("MSILO: no offline messages dumped\n"); } exit; } }

3 2

msilo patch
by Stoyan Mihaylov 07 Feb '13

07 Feb '13

When I used, last versions from git, I got error and crash of kamailo using m_dump. I got latest version from git, and I made one patch: diff --git a/modules/msilo/msilo.c b/modules/msilo/msilo.c index 209b309..4bda8fa 100644 --- a/modules/msilo/msilo.c +++ b/modules/msilo/msilo.c @@ -1125,8 +1125,16 @@ static int m_dump(struct sip_msg* msg, str* owner_s) memcpy(tmp_extra_hdrs.s, extra_hdrs_str.s, extra_hdrs_str.len); memcpy(tmp_extra_hdrs.s+extra_hdrs_str.len, str_vals[4].s, str_vals[4].len); } else { - tmp_extra_hdrs.len = 0; - tmp_extra_hdrs.s = ""; + tmp_extra_hdrs.len = 1; + if ((tmp_extra_hdrs.s = pkg_malloc(1)) == NULL) + { + LM_ERR("Out of pkg memory"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + LM_ERR("failed to free the query result\n"); + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } + tmp_extra_hdrs.s[0]='\x0'; } hdr_str.len = 1024; if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, The problem is: tmp_extra_hdrs.s = ""; and consequent pkg_free(tmp_extra_hdrs.s); As I see, there are couple of ways to solve problem, one of them is setting tmp_extra_hdrs.s = 0; but then we have warnings, or setting tmp_extra_hdrs.s = 0; and check before pkg_free, but pkg_free is used on couple of places. I am not sure what is the best way to report bugs and offer patches.

2 1

Deleting the expired contacts using the Rtimer module
by kiran bhosale 06 Feb '13

06 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

2 1

401 message forwarding
by Алексей Ёлохов 06 Feb '13

06 Feb '13

Hi! I am setting up Kamailio as SBC with upper registration. I can forward REGISTER messages to my softswitch, but 401 UNAUTHORIZED from it is not coming to my UAC. Config section for registration route: route[REGISTER_MY] { rewritehost("3.3.3.3"); t_relay(); } What I see with ngrep (1.1.1.1 - my external UAC ip, 2.2.2.2 - Kamailio ip, 3.3.3.3 - mera softswitch ip): U 1.1.1.1:5060 -> 2.2.2.2:5060 REGISTER sip:2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;rport;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. Max-Forwards: 70. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. User-Agent: SFLphone. Contact: <sip:101@10.2.1.35:5060>. Expires: 600. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. . U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.2.1.35:5060 ;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d;received=1.1.1.1. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=2fb3028f366408eaa444e96dfd845447.8865. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. Contact: <sip:101@10.2.1.35:5060>;expires=600. Server: kamailio (3.3.3 (x86_64/linux)). Content-Length: 0. . U 2.2.2.2:5060 -> 3.3.3.3:5060 REGISTER sip:3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK929f.8339e446.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. Max-Forwards: 69. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. User-Agent: SFLphone. Contact: <sip:101@10.2.1.35:5060>. Expires: 600. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. . U 3.3.3.3:5060 -> 2.2.2.2:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK929f.8339e446.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=d67d5778704611e2a3c600505691470a. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. WWW-Authenticate: Digest realm="SIP-REGISTRAR", nonce="475a20c7". Server: MERA MVTS3G v.4.3.0-38t. Content-Length: 0. . U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=d67d5778704611e2a3c600505691470a. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. WWW-Authenticate: Digest realm="SIP-REGISTRAR", nonce="475a20c7". Server: MERA MVTS3G v.4.3.0-38t. Content-Length: 0. . Thanks in advance! --- Alex Elokhov

1 0

[Fwd: Deleting the expired contacts using the Rtimer module]
by kiran bhosale 06 Feb '13

06 Feb '13

1 0

Deleting the expired contacts using the Rtimer module
by kiran bhosale 06 Feb '13

06 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

1 0

Fwd:
by jose luis 05 Feb '13

05 Feb '13

http://www.lacucinalibanese.altervista.org/rjugo4.php?s=ot

2 1

Syslog
by Vitaliy Aleksandrov 05 Feb '13

05 Feb '13

Hi list. I have a strange problem with syslog. Kamailio doesn't send any messages to syslog from UDP workers that are bound to sock=127.0.0.1:5060. Neither LM_DBG() from a source code nor xlog("any level", "") works. I use kamailio-3.2.4 with this opions: log_stderror=no debug=3. With log_stderror=yes kamailio writes all debug messages to stderr without any problems.

2 4

modifying global data-structures using Rtimer module
by Kiran Bhosale 05 Feb '13

05 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

2 1

Refreshing the list of submitted patches
by Daniel-Constantin Mierla 05 Feb '13

05 Feb '13

Hello, following some discussion over the weekend at Fosdem, it seems that some of submitted patches got forgotten in the mailing list archives and even on tracker. If any of you submitted patches was not accepted or rejected for a persistent reason, please write again to sr-dev mailing list referring to it. It is a short period still good for getting them in v4.0. In general, if there is no more activity regarding a patch for one or two weeks, come back and ask for an update. We don't mind reminders (especially because many of us do have a heavy loaded traveling schedule). Cheers and happy testing upcoming v4.0, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Best way to 'detect' direction
by Paul Belanger 04 Feb '13

04 Feb '13

Greetings, I'm in the process of adding logic to 'detect' the direction of my message, I have to set: rtpproxy_manage("ei") or rtpproxy_manage("ie") respectively. Rather then hard-coding IP addresses in my configuration, I was curious how other people over come this? Perhaps tagging their call control in some other method? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 1

50000+ subscribers on Kamailio?
by Алексей Ёлохов 04 Feb '13

04 Feb '13

Hi! I am choosing a softswitch to serve more than 50000 (up to 100000) active subscribers (each subscriber is using SIP registration). Please tell me , I want to understand can kamailio work with such load? --- Alex Elokhov

2 1

Remove dash (-) from called number
by Ali Jawad 04 Feb '13

04 Feb '13

Dear All I need to remove any dashes wherever they do occur in a called number, any guidance please ? Regards * *

2 3

Re: [SR-Users] sr-users Digest, Vol 92, Issue 80
by Ian French 04 Feb '13

04 Feb '13

Thank you very much Sammy that worked Ian On 29 January 2013 18:14, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Asterisk and dispatcher (Ian French) > 2. Re: Asterisk and dispatcher (SamyGo) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 29 Jan 2013 17:55:40 +0000 > From: Ian French <fretec(a)gmail.com> > Subject: Re: [SR-Users] Asterisk and dispatcher > To: sr-users(a)lists.sip-router.org > Message-ID: > <CAOAGGZSr_24VDL8508XJo9LjkvR= > uo67JoM_MPPyd3bVbWxC7Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 24 January 2013 08:16, <sr-users-request(a)lists.sip-router.org> wrote: > > > Send sr-users mailing list submissions to > > sr-users(a)lists.sip-router.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > or, via email, send a message with subject or body 'help' to > > sr-users-request(a)lists.sip-router.org > > > > You can reach the person managing the list at > > sr-users-owner(a)lists.sip-router.org > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of sr-users digest..." > > > > > > Today's Topics: > > > > 1. Re: Asterisk and dispatcher (SamyGo) > > 2. Re: Asterisk and dispatcher (SamyGo) > > 3. Kamailio Multi tenant as registrar server ony (andrea mucci) > > 4. help me about siremis. (???) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Wed, 23 Jan 2013 22:26:26 -0800 > > From: SamyGo <govoiper(a)gmail.com> > > Subject: Re: [SR-Users] Asterisk and dispatcher > > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > > Users Mailing List" <sr-users(a)lists.sip-router.org> > > Message-ID: > > < > > CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ(a)mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > Hi, > > > > You've metioned this already in your code: > > > > if(ds_is_from_list("1")) > > > > Is this not working for you? > > > > Regards, > > Sammy > > > > > > > > On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec(a)gmail.com> wrote: > > > > > Hi, > > >> I've been working my way through this tutorial ( > > >> > > > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb > > ) > > >> and all it working well thanks. In the other benifits section you > > mention > > >> that this configuration can be used with multiple instances of > Asterisk. > > >> I've configured my [TOASTERISK] route > > > > > > > > > route[TOASTERISK] { > > > ds_select_dst("1","4"); > > > $sht(forw=>$ft)=$du; > > > sl_send_reply("100","Trying"); > > > xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk > > > Cluster [$rm:$au].\n"); > > > route(RELAY); > > > exit; > > > > > > How can i achieve the same for my [FROMASTERISK] route as now with > > > > > > asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" > > > asterisk.bindport = "5060" desc "Asterisk Port" > > > > > > and > > > > > > route[FROMASTERISK] { > > > if($si==$sel(cfg_get.asterisk.bindip) > > > && $sp==$sel(cfg_get.asterisk.bindport)) > > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] > Asterisk > > > Cluster To [$rm:$au].\n"); > > > return 1; > > > return -1; > > > } > > > > > > This allows calls only from cfg_get.asterisk.bindip (single gateway) > > where > > > I would like gateways in the dispatcher list to be selected and marked > > as a > > > positive match > > > > > > if(ds_is_from_list("1")) > > > $sht(forw=>$ft)=$si; > > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] > Asterisk > > > Cluster To [$rm:$au].\n"); > > > return 1; > > > return -1; > > > } > > > > > > Can you help? > > > > > > Thanks in advance > > > Ian > > > _______________________________________________ > > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > > sr-users(a)lists.sip-router.org > > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > >

1 0

Voip Issue
by BrianSuper7 04 Feb '13

04 Feb '13

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hi I have been using your excellent Voip facility for sometime. Recently I have encountered that when I called from 'phone A' to 'Phone B' connects and can hear Phone B's voice but can't hear phone A's voice when dialed from phone A. But when Phone B made the call to phone A, can hear both parties voices. This problem started very recently. I haven't done any configuration changes. This problem incurred nearly a week or two ago. I experienced same problem long time ago and it resolved after few days time. But this time it didn't resolved. My SIP no 87415661. I would appreciate if you could kindly help me to rectify the problem. [...] Content analysis details: (5.2 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 RCVD_IN_DNSWL_NONE RBL: Sender listed at http://www.dnswl.org/, no trust [220.233.0.17 listed in list.dnswl.org] 0.0 FREEMAIL_FROM Sender email is commonly abused enduser mail provider (briansuper7[at]yahoo.com.au) 0.2 FREEMAIL_REPLYTO_END_DIGIT Reply-To freemail username ends in digit ("briansuper7" <briansuper7[at]yahoo.com.au> ) 0.2 FREEMAIL_ENVFROM_END_DIGIT Envelope-from freemail username ends in digit (briansuper7[at]yahoo.com.au) 0.0 HTML_MESSAGE BODY: HTML included in message 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4955] 0.6 REPTO_QUOTE_YAHOO Yahoo! doesn't do quoting like this 3.3 AXB_XMAILER_MIMEOLE_OL_4379D AXB_XMAILER_MIMEOLE_OL_4379D The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

1 0

issue with #<number> calls
by Ivan Milivojevic 03 Feb '13

03 Feb '13

Hi Andrew, Here is one call with special characters: call-id = 100c8a98-c0a8016b-13c4-50029-203fa-68e78324-203fa How we can block such attempts, by rewrite rules? Regards Ivan

1 0

Build 3.3 on ubuntu 12.10
by Gareth Rylance 01 Feb '13

01 Feb '13

Hi I am trying to build the kamailio deb packages on ubuntu 12.04. when I run make deb I get dpkg-checkbuilddeps: Unmet build dependencies: libmono-dev but libmono-dev is not available on 12.04 how do I 1. Fix the dependancy 2. Ignore it as I am not using app_mono (although I have an idea that may in the future) Thanks Gareth

2 2

loose_route causing loop
by Renan Capaverde 31 Jan '13

31 Jan '13

Hello, I have a problem with my configuration file. When I receive a in-dialog request (REINVITE) with topmost route being my kamailio server, it enters in loop and then the call is dropped. What is the easiest way to prevent this to happen? Best Regards, -- *Renan Capaverde* /Estagiário Nível Superior/ /DDT - STE - Soluções em Tecnologia e Embarcados/ *DÍGITRO TECNOLOGIA* *E-mail:* renan.capaverde(a)digitro.com.br <mailto:renan.capaverde@digitro.com.br> *Fone:* +55 48 3281-7000*Ramal:* 8132 *Fax:* +55 48 3281-7299 *Site:* www.digitro.com <http://www.digitro.com> /"Antes de imprimir, pense na sua responsabilidade e no seu compromisso com o meio ambiente"/ Esta mensagem, incluindo seus anexos, é reservada somente à Dígitro e ao destinatário da mensagem. Caso você tenha recebido esta mensagem por engano, queira por favor, retorná-la ao remetente e apagá-la de seus arquivos.

2 1

Re: [SR-Users] Error searching $keepalive.socket pvar in NAT Traversal module
by vito 31 Jan '13

31 Jan '13

Hello, I have: # Kamailio (OpenSER) SIP Server v3.3 # Kamailio (OpenSER) SIP Server v3.0 Defining pv in this format works well: $fs = $(keepalive.socket($du)); Thanks a lot !! Vito

4 4

BasicKamailio DB question
by Jason 31 Jan '13

31 Jan '13

Hi all, excuse the basic nature of my question, I'm just starting out. I've read in a tutorial that it is possible to connect from asterisk to Kamailio and for asterisk to use the Kamailio db for authentication. However, I also read that it is possible to connect to astereisk from Kamailio for retrieving auth data. My question is this... If Kamailio uses the db on a remote asterisk server to retrieve usernames and password info how do you set Kamailio to use multiple databases as you may ultimately have multiple asterisk servers? As far as I could see there is only 1 db typically in the config file. Also, which is the most common configuration and why would one choose one over the other? Thanks in advance.

2 1

REGISTER OK but on send "407 Proxy Authentication Required" with no ACK
by Carsten Maass 30 Jan '13

30 Jan '13

Hi all, I am trying to set up a fax gateway in the following way: PSTN-GW (10.1.1.150) <--> Kamailio (10.1.1.148:5123) <--> t38modem (10.1.1.148:6050) <--> Hylafax PSTN-GW is a standalone Berofix appliance and Kamailio 3.3.3, t38modem 1.2 and Hylafax 6.0.3 running on the same host under Redhat EL6. I used the default kamailio.cfg and just adjusted the PSTN route pattern to "if(!($rU=~"^(\+|00|0)[1-9][0-9]{3,20}$"))", to route all calls starting with 0 to the PSTN-GW. Both PSTN-GW and t38modem Successfully register to Kamailio but when I try to send out a fax from t38modem, Kamailio doesn't ACK the Proxy-Authorization request and t38modem terminates the call: <siptrace> 05:49:11.158157 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 986) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 958 E.....@.@.Y. .d. .d.........INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> Date: Tue, 29 Jan 2013 04:49:11 GMT CSeq: 1 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport User-Agent: T38Modem/1.2.0 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local Organization: Vyacheslav Frolov To: <sip:030987654321@10.1.1.148:5123> Contact: <sip:030123456789@10.1.1.148:6050> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Content-Type: application/sdp Content-Length: 262 Max-Forwards: 70 v=0 o=- 1359434951 1 IN IP4 10.1.1.148 s=Opal SIP Session c=IN IP4 10.1.1.148 t=0 0 m=audio 5008 RTP/AVP 8 101 100 a=sendrecv a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 05:49:11.158927 IP (tos 0x10, ttl 64, id 28825, offset 0, flags [none], proto UDP (17), length 584) 10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556 E..Hp...@.*. .d. .d......4.oSIP/2.0 407 Proxy Authentication Required CSeq: 1 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport=6050 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580 Proxy-Authenticate: Digest realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W" Server: kamailio (3.3.3 (x86_64/linux)) Content-Length: 0 05:49:11.160273 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 501) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 473 E.....@.@.[. .d. .d.........ACK sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> CSeq: 1 ACK Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580 Content-Length: 0 Max-Forwards: 70 05:49:11.166542 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 1195) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 1167 E.....@.@.Y. .d. .d.........INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> Date: Tue, 29 Jan 2013 04:49:11 GMT CSeq: 2 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport User-Agent: T38Modem/1.2.0 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local Organization: Vyacheslav Frolov To: <sip:030987654321@10.1.1.148:5123> Contact: <sip:030123456789@10.1.1.148:6050> Proxy-Authorization: Digest username="979", realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W", uri="sip:030987654321@10.1.1.148:5123", algorithm=MD5, response="48d896ef41a71720b29c56230858582f" Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Content-Type: application/sdp Content-Length: 262 Max-Forwards: 70 v=0 o=- 1359434951 1 IN IP4 10.1.1.148 s=Opal SIP Session c=IN IP4 10.1.1.148 t=0 0 m=audio 5008 RTP/AVP 8 101 100 a=sendrecv a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 05:49:11.167573 IP (tos 0x10, ttl 64, id 28826, offset 0, flags [none], proto UDP (17), length 584) 10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556 E..Hp...@.*. .d. .d......4.oSIP/2.0 407 Proxy Authentication Required CSeq: 2 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport=6050 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.263a Proxy-Authenticate: Digest realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W" Server: kamailio (3.3.3 (x86_64/linux)) Content-Length: 0 05:49:11.169220 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 710) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 682 E.....@.@.Z. .d. .d.........ACK sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> CSeq: 2 ACK Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.263a Proxy-Authorization: Digest username="979", realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W", uri="sip:030987654321@10.1.1.148:5123", algorithm=MD5, response="3056b266b15a92f5cc8cc8e1c073fbcb" Content-Length: 0 Max-Forwards: 70 </siptrace> What do I have to adjust to make this work? Any help and pointers highly appreciated. Thanks in advance and greetings, Carsten. -- Blinkenlichten (Maass & Sacha GbR) - Open Source Solutions Weigandufer 45 - 12059 Berlin - http://www.blinkenlichten.de FON: ++49 +30 13896247 - MAIL: cm(a)blinkenlichten.de FAX: ++49 +30 13896249 - PGP: Key Id 0x2CBCA806 St.Nr. 16/274/61636

2 7

Re: [SR-Users] Asterisk and dispatcher
by Ian French 29 Jan '13

29 Jan '13

On 24 January 2013 08:16, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Asterisk and dispatcher (SamyGo) > 2. Re: Asterisk and dispatcher (SamyGo) > 3. Kamailio Multi tenant as registrar server ony (andrea mucci) > 4. help me about siremis. (???) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 23 Jan 2013 22:26:26 -0800 > From: SamyGo <govoiper(a)gmail.com> > Subject: Re: [SR-Users] Asterisk and dispatcher > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > < > CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > You've metioned this already in your code: > > if(ds_is_from_list("1")) > > Is this not working for you? > > Regards, > Sammy > > > > On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec(a)gmail.com> wrote: > > > Hi, > >> I've been working my way through this tutorial ( > >> > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb > ) > >> and all it working well thanks. In the other benifits section you > mention > >> that this configuration can be used with multiple instances of Asterisk. > >> I've configured my [TOASTERISK] route > > > > > > route[TOASTERISK] { > > ds_select_dst("1","4"); > > $sht(forw=>$ft)=$du; > > sl_send_reply("100","Trying"); > > xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk > > Cluster [$rm:$au].\n"); > > route(RELAY); > > exit; > > > > How can i achieve the same for my [FROMASTERISK] route as now with > > > > asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" > > asterisk.bindport = "5060" desc "Asterisk Port" > > > > and > > > > route[FROMASTERISK] { > > if($si==$sel(cfg_get.asterisk.bindip) > > && $sp==$sel(cfg_get.asterisk.bindport)) > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk > > Cluster To [$rm:$au].\n"); > > return 1; > > return -1; > > } > > > > This allows calls only from cfg_get.asterisk.bindip (single gateway) > where > > I would like gateways in the dispatcher list to be selected and marked > as a > > positive match > > > > if(ds_is_from_list("1")) > > $sht(forw=>$ft)=$si; > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk > > Cluster To [$rm:$au].\n"); > > return 1; > > return -1; > > } > > > > Can you help? > > > > Thanks in advance > > Ian > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > >

2 1

Re: [SR-Users] Default 3.3 config and dispatcher
by Daniel Tryba 29 Jan '13

29 Jan '13

On Tuesday 29 January 2013 16:58:08 Paul Belanger wrote: > > [kamailio] > > host=192.168.1.1 > > context=incoming > > type=friend > > allow=all > > qualify=yes > > insecure=port,invite > > nat=no > > canreinvite=yes > > > > Nothing more to it. > > Thanks for pointing me in the right direction. Will provide feedback > shortly. One more thing I might have forgotten. Asterisk needs to responde with a 200 OK to the OPTIONS requests in order for Kamailio to think the dispatcher route is up and runnnig. So add an s exten in the asterisk context: [incoming] exten => s,1,NoOp() -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

1 0

Subscribe and PUA database
by Misha Ketslah 29 Jan '13

29 Jan '13

Hello, i'm trying to set BLF on my snom 320 phone to get notification on function key for certain extension as well registered against kamailio. Any other info i can provide except the following one and as well i want to exuse me, as i'm pretty noob to all this, and my question might be silly, but after hours in front of google i found no solution :) I always get that: after pua_set_publish() ** 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=780 a=41 n=isflagset 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=16 n=if 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=save 3(7825) ERROR: pua [send_publish.c:578]: New PUBLISH and no body found- invalid request 3(7825) ERROR: pua_usrloc [ul_publish.c:326]: while sending publish 3(7825) ERROR: pua_usrloc [ul_publish.c:327]: TEST: type & UL_CONTACT_UPDATE is 2, and error is -1 (^^^ This line was added by me into module as i saw related thread and the patch solution, though it never meets condition and fails to insert from there as well.) 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=2 n=exit my configs are : in BLF i set <sip:666@SERV_IP;user=phone> in cfg i have alias="SERV_IP" listen=udp:SERV_IP port=5060 loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "speeddial.so" ... modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) ... modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode" , 3 ) #modparam("presence", "publ_cache", 0) modparam("presence", "expires_offset", 300) modparam("presence", "max_expires", 3600) #modparam("presence", "force_active", 1) modparam("presence", "db_update_period", 10) modparam("presence", "server_address", "sip:SERV_IP:5060") modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("pua", "db_url" ,DBURL ) modparam("pua", "db_mode", 2) modparam("pua", "db_table", "pua") #modparam("pua", "outbound_proxy", "sip:SERV_IP") modparam("pua", "min_expires", 10) modparam("pua", "default_expires", 3600) modparam("pua", "update_period", 60) modparam("pua_usrloc", "default_domain", "SERV_IP") request_route { if(method=="NOTIFY") { xlog(" pua_update_contact $rm from $fu (IP:$si:$sp) \n"); if(!pua_update_contact()) xlog("pua update failed \n"); } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); route(SPEEDDIAL); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); route(DISPATCH); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("pua_set_publish()\n"); if(!pua_set_publish()) xlog("set publish failed \n"); xlog(" after pua_set_publish()\n"); if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #pua_set_publish(); if (!save("location")) { sl_reply_error(); xlog("save location failed \n"); } exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL if(uri=~"sip:[0-9]{2}@.*") xlog("speeddials\n"); # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") sd_lookup("speed_dial"); route(SPEEDDIAL); xlog("PUBLISH|SUBSCRIBE"); sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; append_to_reply("Contact: <sip:82.80.18.100:5060>\r\n"); if(is_method("PUBLISH")) { xlog("handle publish here\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle subscribe here \n"); handle_subscribe(); t_release(); } exit; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { ....... return; } # XMLRPC routing route[XMLRPC] { .... } #!endif # route to voicemail server route[TOVOICEMAIL] { ...... return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif } route[SPEEDDIAL] { ..... return; } ....

1 0

Web softphone using websockets against Kamailio
by Pirjo Ahvenainen 29 Jan '13

29 Jan '13

Greetings gurus! I'm playing with an idea to create a web based softphone (html5 + no installations for the end user) and use Kamailio's websocket module for backend. I'd love to hear about your comments, challenges and successes using such configuration. Is it a feasible way to construct a softphone even today when even IE9 does not support websockets, as such? I'm sure IE9 will end up in specs as a must-support platform. A collegue tried using sipml5 with webrtc against a SnomONE pbx (I know... ;)), and said there's no way it can work, but I'm not convinced the idea itself wouldn't work. It would help me lots if I could make a simple example using Kamailio with SIP over websockets, can you comment on how much effort do I need on Kamailio side to make this work? Do I need off-default config scripting, or is it enough to just set up the module and set the parameters? And even with the risk of stepping a little off topic, if anyone has worked on web based softphones, I'd love to hear if you can recommend on how to approach this. Cheers, Pirjo --

2 3

Default 3.3 config and dispatcher
by Paul Belanger 29 Jan '13

29 Jan '13

Greetings List, First post of many. I've managed to get up a running a very simple VoIP network with Kamailio as the core, and asterisk on the edge. Most of this has been accomplished using the default kamailio.cfg file shipped in 3.3. For my next adventure, I'd like to start using the dispatch module to route calls from Kamailio to asterisk, however having some trouble getting my syntax correct. I was hoping somebody could point me to an existing post / URL describing what I was asking, if it is based on the default configuration file, the better. I should note, if I have Asterisk register to kamailio, I can route calls fine however, I believe the dispatcher is the better way to do this. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 1

Major problem with setup NAT + MULTIDOMAIN
by Rumen Mihailov 29 Jan '13

29 Jan '13

Hi guys, I've baning my head for the last 6 7 days with something that have to be universally simple but obviously for me it is not, so please do help. I have the following setup. xx.xx.xx.xx (real IP 1:1 NAT) -> 10.2.47.201 Kamailio listens on 10.2.47.201 I have setup a domain xxx.itradebg.com and I've made the appropriate records in the DNS. I have inserted the domain xxx.itradebg.com in the domain table in the database and kamctl domain show shows the correct thing. Now when I try to register 999(a)xxx.itradebg.com I get Request timed out I have this in my config: if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "sip", "1")) { auth_challenge("$fd", "0"); exit; } } if (from_uri!=myself && uri!=myself) { if(lookup_domain("$fd", "@from.uri.host")){ xlog("LOCAL DOMAIN -> REGISTER"); exit; } else{ xlog("L_WARN","FROM NOT LOCAL"); sl_send_reply("403","Sorry mate not relaying"); exit; } } By the time of writing this I realised that this should be all rewritten to this only, as the second check is useless: if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "sip", "1")) { auth_challenge("$fd", "0"); exit; } } However I get 401 Unauthorized. The previous setup was giving me 408 - Request Timed out... I am pretty sure I have the passwords correct. What I cannot understand is why authentication fails ?! Also, how do I set kamailio to print debug output ? debug=9 log_stderror=no This is what I have in the .conf file. /var/log/messages seems to only have what I have printed with xlog... (think) I am really out of ideas... perhaps I need some sleep. Anyway any input is helpful guys. Thanks, Rumen http://itradebg.com

2 4

Error searching $keepalive.socket pvar in NAT Traversal module
by vito 28 Jan '13

28 Jan '13

Hello, I started to use Kamailio few months ago. Maybe this is a stupid issue but I can´t start Kamailio with NAT TRAVERSAL module and $keepalive.socket pseudo-variable. My script configuration is the default template adding this lines (extracted from nat traversal doc): ... loadmodule "nat_traversal.so" ... ... $fs = $keepalive.socket($du); ... I have this error: -e 0(17455) ERROR: <core> [pvapi.c:501]: error searching pvar "keepalive" 0(17455) ERROR: <core> [pvapi.c:705]: wrong char [e/101] in [$keepalive] at [9 (0)] 0(17455) : <core> [cfg.y:3330]: parse error in config file /etc/kamailio/kamailio.cfg, line 410, column 8-17: unknown script pseudo variable $keepalive ERROR: bad config file (1 errors) I tried in this environments: Ubuntu 10.04.4 LTS x86_64 Kamailio 3.3.3+git20130128+lucid1 Ubuntu 10.04.4 LTS i686 Kamailio 3.0.4+lucid1 Ubuntu 12.04.1 LTS x86_64 Kamailio 3.3.2+precise1 All packages instaled from deb repositories (http://deb.kamailio.org/) I think there are problem parsing "$keepalive.socket" pseudovariable, because Kamailio doesn´t scape this character "." and it doesn´t recognise the pseudovar. Does anyone have this problem? Regards

2 1

Sip express live Cd router
by Shoukat, Kashif 28 Jan '13

28 Jan '13

Dear , Can you please provide me link of sip express live Cd router .. I want to use it... waiting your kind response Regards, Kashif Shoukat | Analyst Software Engineer | Work : + 1.202.289.9898 x 9077 Mobile : + 92.323.431.8548 Skype: kashif.shoukat.dgs www.dgsworld.com<http://www.dgsworld.com/> dgs deliberate by design This message contains information which may be confidential and privileged. Unless you are the addressee (or authorized to receive for the addressee), you may not use, copy or disclose to anyone the message or any information contained in the message. If you have received the message in error, please advise the sender by replying to kashif.shoukat(a)dgsworld.com<mailto:kashif.shoukat@dgsworld.com> and delete the message.

1 0

TLS certificate chain Android verification
by Nord7 28 Jan '13

28 Jan '13

Hello all. Our company buy Thawte SSL Web server certificate. I'm enabled in Kamailio 3.3 TLS support and configured him. Adding to "myhost.pem" certificate, two "intermediate" certificates (like in chain) "cat myhost.pem SSL_SecondaryCA.pem SSL_PrimaryCA.pem > chain-cert.pem". Remark: I use this chain on apache2. And Android trusted to this certificate chain. After this i use chain-cert.pem with KAMAILIO 3.3 and SIP client "BRIA Android". "Bria Android" says certificate is not trusted. Next step "BRIA IPHONE" and certificate chain works (trusted)! BRIA support says me - BRIA don't trusts certificate if system don't trusts. Why Android don't trusts to my chain-certificates with KAMAILIO? If his trust to same certificate with Apache? I'm understand this is not Kamailio problem, but maybe any one who have expirience in this problem help me? I know about bouncycastle, but this is not best decision for me. Maybe something else can help? I tryed this with Android 2.3.7, 4.0, 4.1. Best regards Kirill.

1 0

event_route for pre-dialog start
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I think that it is more than likely not implemented, but is there any event-route that is triggered just before event_route[dialog:start] ? I need to check some security flags before the dialog is created. But it is too early too check them in relay route.. Mino

1 0

Is kamailio creating multiple connections to db?
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I made some DB test on our HA cluster, and I got that for 1 process I have 1000 queries/sec and for 11 processes 3000 queries/sec. That's why I would to ask if every kamailio process that is spawned has its own database connection or queries are put in the queue and executed by some main process ? I would like to know this, so that I could assess maximum call count/sec that db can handle. Thank you, Mino

3 4

Generating logging based on SIP messages
by Grant Bagdasarian 28 Jan '13

28 Jan '13

Hello, A while ago I came across the siptrace and sipcapture modules and thought this would be a good way to generate logging out of all the SIP messages that are passed through Kamailio and stored in the siptrace/sipcapture table. My idea was to extract certain values from SIP messages and store these in our database. Also each SIP Request or Response would be mapped to a certain status so we would be able to monitor which call is in which state. I still have a lot of work to do to think this all out, since it's still only an idea. I'm just wondering if anyone could give me some pointers into what to keep in mind when doing this. Regards, Grant

2 2

Pseudo variable to get instance in the contact....
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, I added in my local branch "$ct.inst" pseudo variable to retrieve instance. Since instance could be an important attribute to make routing decisions, having quick access to it in the message may be useful. Anyone else need this functionality? Krish Kura

3 2

iptel x media5-fone problem
by Kasper Hämäläinen 28 Jan '13

28 Jan '13

hi guys, hope u got time to help me. Im first timer testing VoIP and SIP. I created account to ur iptel.org site. I downloaded "Jitsi"-software. First when I opened it, I login with my username and password. Then I go to "add new account", give same login username and password, but Jitsi says that my password is wrong..wtf. And I installed Media5-Fone to my iPhone 4S. When adding SIP-account it's says again that I got wrong password. And YES, I didn't mistype it. Thanks - Kasper

2 1

CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg - Kamailio 3.3.x
by Rinor Hoxha 28 Jan '13

28 Jan '13

Hi list, I'm having some issues with dialog module. From time to time I get the following error: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | sort | uniq -c 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) This one was flooding my log file: tl=0x2aeeea1c6840 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this happening and how to fix it. (I suspect that may be some issues with our server memory...checking now...and will inform). However wanted to raise this issue in case anyone else is having the same. Thanks, Rinor

2 1

Pseudo variable to access SIP instance in contact
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, Is there a way to directly access the SIP instance present in message from the script? Thanks Krish Kura

2 1

Kamailio behind NAT - best practice
by Klaus Darilion 27 Jan '13

27 Jan '13

Hi! Up to now I could avoid Kamailio setups with Kamailio behind NAt. But this time I have to deal with it. That's why I want to ask what others did as best practice. The scenario is: Asterisk 1----\ Kamailio+RTPPROXY \ |10.10.0.2 Asterisk n------\ | --------|------------- FW --SIP-trunk--- ITSP Freeswitch 1----/ 10.10.0.1 public-IP / Freeswitch n--/ 10.10.0.x Kamailio and rtpproxy have a private IP. Internal communication uses private IPs, external communication uses a public IP which is NATed 1:1 to Kamailio's IP address. No registrations, just forwarding of messages. Using the global "advertised_address" setting with the public IP does not work, as there is also internal communication. Using set_advertised_address() is also cumbersome. So it seems, the easiest solution would be to use 2 sockets on Kamailio, e.g. port 5050 and port 5060. Then I could use the listen with dedicated advertised addresses: listen=udp:10.10.0.2:5050 listen=udp:10.10.0.2:5060 advertise pu.bl.ic.ip:5060 If I understand it correctly, this should solve all issues with Record-Routing and Via-headers. For RTP-Proxy it seems necessary to detect the direction of each message and set the IP address in rtpproxy_manage("","ip.add.re.ss") manually. Thus, it seems straight forward - or do I miss something? Any comments and practical experience? Thanks Klaus

5 8

Pseudo variables and uid_domain module compatibility
by Philippe Sultan 25 Jan '13

25 Jan '13

Hi, According to docs of "uid_domain.so", $fd and $td are attribute groups which are set when calling the lookup_domain function, to retrieve domain attributes stored in an DB ($fd.did or $td.did) : lookup_domain("$fd", "@from.uri.host"); lookup_domain("$td", "@ruri.host"); if (strempty($fd.did) && strempty($td.did)) { # Neither the calling nor the called domain is local # This is a relaying attempt which should be forbidden sl_reply("403", "Relaying Forbidden"); drop; } But then, starting Kamailio fails with a configuration file error : 0(2098) ERROR: <core> [pvapi.c:761]: error searching pvar "fd" 0(2098) ERROR: <core> [pvapi.c:965]: wrong char [d/100] in [$fd] at [2 (0)] 0(2098) : <core> [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 332, column 15-17: unknown script pseudo variable $fd ERROR: bad config file (1 errors) Line 332 is : if (strempty($fd.did) && strempty($td.did)) { I thought I could start Kamailio without "pv.so" (and use selects instead), but the error still shows up, as it is reported by the core pvapi.c file. So I'm kind of stuck, and asking help from the community to fix this problem :D Philippe

2 8

uac_replace_from recovery with modified header
by Andreas Granig 25 Jan '13

25 Jan '13

Hi, This is a question for kamailio 3.0.x and I'm still trying to reproduce this issue somehow for latest stable, but maybe it was a known issue anyways which got fixed in newer versions, so I don't have to dig deeper into this: There is an INVITE "A -> kamailio -> B", and kamailio does uac_replace_from() to replace the display- and user-part (leaving the domain-part untouched). When there's a BYE "B -> kamailio -> A", then From/To are switched, so the replaced From of the INVITE becomes the replaced To of the BYE. Kamailio gets that one right and tries to recover (in auto-mode) the To. However, in this BYE, B changes the domain-part of the To, which should be valid, as elements are only supposed to inspect the tags of From/To, right? The problem though is that when recovering the To, kamailio inserts garbage into the domain part. The vsf-Parameter in the Route of the BYE is definitely ok though, so that doesn't seem to be the issue. Here's an example: In the INVITE, From: "ab12345"<sip:ab12345@192.168.0.12;user=phone>;tag=948a3340 gets rewritten to From: "12345"<sip:12345@192.168.0.12;user=phone>;tag=948a3340 And in the BYE, To: "12345"<sip:12345@1.2.3.4>;tag=948a3340 gets rewritten to To: "12345"<sip:ab12345<@-%:68*0.0%12;user=phone>;tag=948a3340 Note that the received To in the BYE has a different host-part than what got sent out in the From of the INVITE, and when kamailio recovers the To, it doesn't recover the Display-part (not sure though if this is intended behaviour anyway), but it only party recovers the domain-part. You can still see some artefacts of the original domain (like the "68" from the second "168" octet, and the "12" from the last octet). Has anyone encountered this issue before? Andreas

4 12

help me about siremis.
by 문종민 25 Jan '13

25 Jan '13

I'm developers who trying to install siremis. will go in there and install siremis website. However, server-services -> domain list by pressing will get the following error. What can I do to fix this?

2 1

problem with db access
by MASebastian 25 Jan '13

25 Jan '13

Hi everyone, Recently I work on kamailio's snmpstats module. I'm trying to retrieve some data from database, the code pass the compilation but when starting kamailio always get this error. Jan 25 10:57:05 localhost kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/local/lib/kamailio/modules_k/snmpstats.so: undefined symbol: db_bind_mod I take the code from http://www.asipto.com/pub/kamailio-devel-guide/#c09database and ursloc module for example. not sure if I still miss something to make db access work. Any suggestion? Thanks, Sebastian

2 1

Websockets WSS problem with NOTIFY
by Pete Kelly 24 Jan '13

24 Jan '13

Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying "Compressed bit must be 0 if no negotiated deflate-frame extension" The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-s… Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point.

3 14

sipcapture + db_postgres gives lots of warnings on PostgreSQL server
by Øyvind Kolbu 24 Jan '13

24 Jan '13

Hi, while trying to use sipcapture together with PostgreSQL I've found that it stores the "\r\n" in SIP messages as "\015\012", though not without a lot of noise in the PostgreSQL log files. For every line inserted in PostgreSQL I get: [0] WARNING: nonstandard use of \\ in a string literal at character 763 [0] HINT: Use the escape string syntax for backslashes, e.g., E'\\'. [0] STATEMENT: insert into sip_capture (date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,msg) values ('2013-01-24 15:27:43',1359037663218925,'OPTIONS','','sip:129.240.253.134','','keepalive','533cb9e91f4b999cf76861cbb9ed54ed-149d','','','','','','5001ed5d-6979(a)127.0.0.1','','SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0','z9hG4bKae6f.d4a02ac7.0','10 OPTIONS','','','','kamailio (3.3.3 (i386/linux))','129.240.253.197',5060,'129.240.253.134',5060,'',0,'',0,1,2,'',1,'voipt1:21','OPTIONS sip:129.240.253.134 SIP/2.0\\015\\012Via: SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0\\015\\012To: sip:129.240.253.134\\015\\012From: sip:keepalive@edgetest.voip.uio.no;tag=533cb9e91f4b999cf76861cbb9ed54ed-149d\\015\\012CSeq: 10 OPTIONS\\015\\012Call-ID: 5001ed5d-6979(a)127.0.0.1\\015\\012Content-Length: 0\\015\\012User-Agent: kamailio (3.3.3 (i386/linux))\\015\\012\\015\\012') Kamailio as of latest git on the 3.3 branc + 3e580e1ee24f9b9ec4b1fa151b23ebc84f3db1cb. Same warnings when linked with both PostgreSQL 8.4.13 and 9.1.1. According to Alexandr Dubovikov MySQL stores '\r\n' as '\r\n'. -- Øyvind Kolbu

1 0

Decode contact with user paramer
by Gareth Rylance 24 Jan '13

24 Jan '13

When using the siputils function encode_contact() on a message that contains a user parameter e.g. <sip:XXXXX00015;phone-context=national@10.YYY.1.92;user=phone>;tag=88930 it encodes fine e.g. <sip:natted_ua*sip:XXXXX00015;phone-context=national**192.168.242.102*5080*udp@10.YYY.70.100 > but decode_contact() errors due to the ";" (there is an explicit check for it in the code) I am currently working round this using $rU = $(rU{s.replace,;,:}); to remove the ; as the user part is not needed for my application. Would it be safe to remove the check for ";" ? Gareth

3 6

Kamailio Multi tenant as registrar server ony
by andrea mucci 24 Jan '13

24 Jan '13

Hi Gurus, i have a basic question. i have a SIP PBX server that don't have the Multi Tenant Feature or Multi Company Feature. i now that Kamailio support this feature, so is possible to use the kamailio as registrar server and redirect all the services, SUBSCRIBE, NOTIFY, and RTP to the SIP PBX with corrects headers? Thanks for your help

3 2

Asterisk and dispatcher
by Ian French 24 Jan '13

24 Jan '13

> > Hi, > I've been working my way through this tutorial ( > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb) > and all it working well thanks. In the other benifits section you mention > that this configuration can be used with multiple instances of Asterisk. > I've configured my [TOASTERISK] route route[TOASTERISK] { ds_select_dst("1","4"); $sht(forw=>$ft)=$du; sl_send_reply("100","Trying"); xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk Cluster [$rm:$au].\n"); route(RELAY); exit; How can i achieve the same for my [FROMASTERISK] route as now with asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" and route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } This allows calls only from cfg_get.asterisk.bindip (single gateway) where I would like gateways in the dispatcher list to be selected and marked as a positive match if(ds_is_from_list("1")) $sht(forw=>$ft)=$si; xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } Can you help? Thanks in advance Ian

2 2

Guidelines to upgrade to 4.0.0
by Daniel-Constantin Mierla 23 Jan '13

23 Jan '13

Hello, the wiki page collecting the guidelines for upgrading from 3.3.x to 4.0.0 is available at: - http://www.kamailio.org/wiki/install/upgrade/3.3.x-to-4.0.0 I just added the SQL statements needed to upgrade the MySQL database, but there is lot of other stuff to be added. Hopefully, anyone noticing changes will add few notes and instructions about how to upgrade. The wiki can be edited if you make an account, it is not required to be a developer to add notes to this page. Would be great if users of postgres will add the sql statements specific for this database type. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Problem with 4.0 nightlies and IMS - sem_post undefined (Barry Flanagan)
by madhumanjusha＠integramicro.com 23 Jan '13

23 Jan '13

On Tue, January 22, 2013 1:58 pm, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > > > > > > > > > Message: 3 > Date: Tue, 22 Jan 2013 13:19:44 +0000 > From: Barry Flanagan <barry(a)flanagan.ie> > Subject: [SR-Users] Problem with 4.0 nightlies and IMS - sem_post > undefined To: sr-users(a)lists.sip-router.org > Message-ID: > <CALJb54-rPPbjgGR5iEKYztNGDvT11znthJzgMYovpp_t6N0S8w(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > > Hi > > > I am trying to run the Ubuntu nightlies from > http://deb.kamailio.org/kamailiodev-nightly and loading the p-cscf config > files. > > Things were working fine while I was on the 20130110 nightlies, but when > I > upgraded to 20130122 I get the following errors. Any have an idea why > sem_post is undefined and how to fix this? > > > > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_registrar_pcscf.so>: > /usr/lib64/kamailio/modules/ims_registrar_pcscf.so: undefined symbol: > sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could > not open module </usr/lib64/kamailio/modules/cdp.so>: > /usr/lib64/kamailio/modules/cdp.so: undefined symbol: sem_post > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_qos.so>: > /usr/lib64/kamailio/modules/ims_qos.so: undefined symbol: sem_post > > > Thanks > > > -Barry Flanagan >

2 1

Modify IP address in SDP
by Austin Einter 23 Jan '13

23 Jan '13

Dear All When 200 OK to INVITE passes through Kamailio proxy , I want to change the IP address in SDP to a specific IP address. Can this be done by just modifying configuration files or do I need to modify in source code. In case I need to modify source code, which files I need to look at. Kindly provide pointers for said problem. Thanks Austin

3 7

FOSDEM
by Peter Dunkley 22 Jan '13

22 Jan '13

Hello, Are there any plans for a Kamailio meal/get together on the Saturday evening this year? Peter

9 19

Patch to resolve segfault in json module
by Richard Brady 22 Jan '13

22 Jan '13

Hi folks If an empty or invalid JSON string is passed to the json_get_field function from the the json module, it causes a segfault. I have attached a patch. Regards, Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

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Re: [SR-Users] Kamailio as a front-end for TLS-UDP
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Do no send replies to private address only, keep mailing list otherwise future such emails will be discarded. The contact has to stay the same, because it needs to be the address of the device. Record-Route and Path are designed to take care of routing back to lb, which will use tls to send to device. Cheers, Daniel On 1/22/13 12:26 PM, Anton Kvashenkin wrote: > Thanks for reply. Yes, kamailio acts as a balancer sending REGISTER > and INVITES, etc. The problem is that at L3-L4 it works well (sends > all datagram messages to 10.0.0.1:5060 <http://10.0.0.1:5060> UDP) but > at L7 the Contact header: <>;transport=tls. > > > 2013/1/22 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 1/22/13 10:56 AM, Anton Kvashenkin wrote: >> Guys, I've already asked @ G+ Kamailio coummunity but had no >> answer so I'm asking here: >> >> I'm trying to deploy +Kamailio >> <https://plus.google.com/114984922071338017631> as a front-end >> for TLS-UDP translation, as a back-end would be +FreeSWITCH >> <https://plus.google.com/104108080382960591162> . So, is kamailio >> so smart to fix Contact or Path header for ;transport=tls or it >> translates only on L3-L4 layer and I should fix those headers >> manually? > is it only for calls or you need to forward registrations to > kamailio as well? > > Everything is done automatically, for calls you just need to do > record routing. For registrations you have to use path module and > be sure that freeswitch supports sip path extension. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > -http://conference.kamailio.com - > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio as a front-end for TLS-UDP
by Anton Kvashenkin 22 Jan '13

22 Jan '13

Guys, I've already asked @ G+ Kamailio coummunity but had no answer so I'm asking here: I'm trying to deploy +Kamailio<https://plus.google.com/114984922071338017631> as a front-end for TLS-UDP translation, as a back-end would be +FreeSWITCH<https://plus.google.com/104108080382960591162> . So, is kamailio so smart to fix Contact or Path header for ;transport=tls or it translates only on L3-L4 layer and I should fix those headers manually?

2 1

VoIP diner at FOSDEM 2013
by Benjamin Henrion 22 Jan '13

22 Jan '13

Hi, I am organizing a diner for all Voice over IP hackers (Asterisk, Freeswitch, Kamailio, OpenSER, Sipvicious :-), etc…) during the Saturday evening of FOSDEM 2013. There is a complete track at the Telephony devroom as well: https://fosdem.org/2013/schedule/track/telephony/ More infos are registration: http://www.zoobab.com/voip-diner-fosdem Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

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Wiki page for what is new in v4.0.0
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Hello, there is a skeleton content for what is new in upcoming v4.0.0, but there are many new features missing in the lists. Please try to add short notes about the features you know they were added for 4.0. The wiki page is available at: * http://www.kamailio.org/wiki/features/new-in-devel Anyone that makes an account can edit: * https://www.kamailio.org/wiki/start?do=register Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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PATH advertised
by Olle E. Johansson 22 Jan '13

22 Jan '13

Hi! If I'm running Kamailio behind NAT and need to add a Path header with the outside IP address and port, that is configured as the "advertise" address - how do I do that? Can I add two Path headers at the same time? I can't find a way to add it like I can with Record-Route headers. /O

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Re: [SR-Users] sr-users Digest, Vol 92, Issue 43
by madhumanjusha＠integramicro.com 22 Jan '13

22 Jan '13

Thanks a lot for quick response. On Sat, January 19, 2013 3:59 am, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Mid call announcement(kamailio server) > (madhumanjusha(a)integramicro.com) > 2. Re: Mid call announcement(kamailio server) > (rabs(a)dimension-virtual.com) > 3. CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus > dlg - Kamailio 3.3.x (Rinor Hoxha) 4. Huge undertaking and in need of > advices from the community (Rumen Mihailov) > 5. Re: Huge undertaking and in need of advices from the > community (Stoyan Mihaylov) 6. Re: Invite - negotiate to asterisk as peer > not kamailio (Scott, Matt) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Fri, 18 Jan 2013 06:52:15 -0500 (EST) > From: madhumanjusha(a)integramicro.com > Subject: [SR-Users] Mid call announcement(kamailio server) > To: sr-users(a)lists.sip-router.org > Message-ID: > <39960.61.8.152.138.1358509935.squirrel(a)mail.integramicro.com> > Content-Type: text/plain;charset=iso-8859-1 > > > Hello, > As of now am working on Kamailio server. > My task is to announce in between call,my sip clients are EKIGA. > I should able to make my kamailio server act as controller and media > server,for that purpose my first step is : invite hold to called party. > second step is:: invite(sdp for .wav file) to caller and then am sending > rtp streaming.In wireshark am able to see rtp packets,but caller is not > playing the .wav file which am streaming through ortp library from > kamailio. > > Can anyone help me out with this problem?On top of it,is this possible in > reality? > > > Thanks & Regards, > Manjusha A. > Integra Micro Software Services (P) Ltd. > > > > > > > > ------------------------------ > > > Message: 2 > Date: Fri, 18 Jan 2013 15:14:49 +0100 (CET) > From: rabs(a)dimension-virtual.com > Subject: Re: [SR-Users] Mid call announcement(kamailio server) > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <38001.212.40.242.42.1358518489.squirrel(a)www.rodriguezfeo.es> > Content-Type: text/plain;charset=iso-8859-1 > > >> Hello, >> As of now am working on Kamailio server. >> My task is to announce in between call,my sip clients are EKIGA. >> I should able to make my kamailio server act as controller and media >> server,for that purpose my first step is : invite hold to called party. >> second step is:: invite(sdp for .wav file) to caller and then am >> sending rtp streaming.In wireshark am able to see rtp packets,but caller >> is not playing the .wav file which am streaming through ortp library >> from kamailio. >> >> Can anyone help me out with this problem?On top of it,is this possible >> in reality? > > No, it's not possible using only kamailio, and thats because kamailio > it's a SIP proxy, it have nothing to do with media, for what you are > triying to get you need to router your calls throught a B2BUA, like sems > or asterisk. > > Best regards > > > > > > ------------------------------ > > > Message: 3 > Date: Fri, 18 Jan 2013 14:11:05 +0100 > From: Rinor Hoxha <rinorhoxha(a)gmail.com> > Subject: [SR-Users] CRITICAL: dialog [dlg_timer.c:205]: Trying to > update a bogus dlg - Kamailio 3.3.x To: sr-users(a)lists.sip-router.org > Message-ID: > <CAFvba4ipKqicMJ0qn9UNLXmhjGhf_83fzG=L5w2qVBpc7JC1GQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi list, > > > I'm having some issues with dialog module. From time to time I get the > following error: > > ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | > sort | uniq -c > > 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) > > This one was flooding my log file: tl=0x2aeeea1c6840 > > > 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this > happening and how to fix it. (I suspect that may be some issues with our > server memory...checking now...and will inform). However wanted to raise > this issue in case anyone else is having the same. > > Thanks, Rinor >

1 0

Kamailio Development Workshop, Feb 14-15, 2013, Alicante, Spain
by Daniel-Constantin Mierla 21 Jan '13

21 Jan '13

Hello, I want to announce the next Kamailio Development Workshop, in Alicante, Spain, during Feb 14-15, 2013, organized by Asipto and University of Alicante, Department of Master Degrees in Telecommunication Engineering. You can find more details at: - http://www.kamailio.org/w/2013/01/kamailio-development-workshop-feb-14-15-2… Like the previous one of its kind (done in 2011 in Barcelona), this type of workshop targets to explain the source code structure and internal APIs of Kamailio, to speed up the learning of how to write own Kamailio extensions in C (e.g., write a module, which will be part also of the content). Its low registration fee is set to cover organizational logistics, the number of seats is limited, participants are accepted in the order of registration. Besides the workshop during the two days, on the evening of Feb 14 is planned a VoIP dinner. You can participate to it even if not coming to workshop. It will be nice to meet face to face and talk about latest trends in RTC/UC. If you need further details, contact me via email. Regards, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

all modules in one folder
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, please note that all modules are now back in one folder, respectively 'modules/'. There were few modules I couldn't compile due to missing exotic libraries, but based on grep, they should not depend on older modules paths. Anyhow, testing that the set of modules you are using compile and load file would be recommended. The folders modules_k and modules_s are now empty, with a placeholder .gitkeep, to be removed later if there is no need to revert any module relocation. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

updates to perl modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, there were several changes pushed to master branch related to perl modules: - perl was renamed to app_perl - to follow the naming pattern for embedded interpreters - embedded perl library is named now Kamailio (it was OpenSER so far, thanks to patches from Stoyan Mihaylov) - perlvdb was renamed to db_perlvd because it is a db driver module If you are using these modules, you will have to update your configuration file and perl scripts. Testing and feedback is appreciated. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

renaming to kamailio in perl and snmpstats modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, I'm writing to see if anyone has some spare time to take care of renaming from openser to kamailio in perl and snmpstats modules. I assume is more like search and replace, but has to be carefully done to be coherent and not break stuff there, so it might be a bit more than a sed command. The IANA enterprise number in MIBs is already the one for kamailio.org Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 7

Testdrive the coming release 4.0 now!
by Olle E. Johansson 19 Jan '13

19 Jan '13

http://www.kamailio.org/w/2013/01/test-4-0-now/ Friends, We do need your help testing 4.0 now. - Compile on your systems, from small embedded systems to large servers, from Linux to *BSD and Solaris - Use with your configuration script - Test the new modules and functions - Read the updated documentation files On the link above, you have instructions on how to build from GIT sources and install a test system. We do look forward to your feedback! Let's all work to make this a very, very good and succesful release! Regards, /O ---- Kamailio on Twitter: @kamailioproject (also on Facebook and Google plus)

1 0

Huge undertaking and in need of advices from the community
by Rumen Mihailov 18 Jan '13

18 Jan '13

Hi all, Thank you for all the help till now. Until now I was trying to setup completely redundant setup with Kamailio + asterisk boxes and for the moment i have the following configuration working. 2x Kamailios (Active/Passive) 2x Asterisks Kamailio is load balancing I can add more boxes if needed MySQL Cluster for all database needs Now the time has come and I will need to made this setup available for around 10K users. I will need to become an ITSP for an Internet Provider that is missing the telephony part from his package, so I have a couple of questions. 1. Is this setup OK to cover the needs of the customer base ? hardware will be decent 2. The end users will most probably need ATA adapters. What is the best cost effective solution I can go for ? 3. Do I need any other piece of software apart from the SIP Proxy and asterisk as I need the cost of implementation to be as low as possible. 4. Pricing...I can see that there are a lot of plans of the competitors that include 1000 minutes for free...Do you have any statistics what is the avarage % usage of prepaid minutes ? 5. Are there any other "stones" that I might hit and should be aware of ? Thank you for the information Best regards, Rumen

2 1

Merry Christmas!
by Daniel-Constantin Mierla 18 Jan '13

18 Jan '13

Another year getting to its end! Looking back, looks like one with the greatest achievements in the development of the project so far. Year is not done, so I save that summary for one week later, there is still stuff on its way to our source code repository. Now I want to thank to everyone promoting and contributing to the project, from developers to community members, and wish Merry Christmas and great winter holidays to all supporters of our project! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

8 8

Mid call announcement(kamailio server)
by madhumanjusha＠integramicro.com 18 Jan '13

18 Jan '13

Hello, As of now am working on Kamailio server. My task is to announce in between call,my sip clients are EKIGA. I should able to make my kamailio server act as controller and media server,for that purpose my first step is : invite hold to called party. second step is:: invite(sdp for .wav file) to caller and then am sending rtp streaming.In wireshark am able to see rtp packets,but caller is not playing the .wav file which am streaming through ortp library from kamailio. Can anyone help me out with this problem?On top of it,is this possible in reality? Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 1

HEAD: Problems with $$ in xlog
by Klaus Darilion 17 Jan '13

17 Jan '13

Hi! In xlog I used to use $$ to escape the dollar sign to not be interpreted as PV, eg: xlog("L_INFO"," $$ru\n"); Current HEAD gives an error: ERROR: <core> [pvapi.c:757]: error searching pvar "" ERROR: <core> [pvapi.c:961]: wrong char [$/36] in [$] at [0 (0)] ERROR: xlog [xlog.c:363]: wrong format[ $$ru#012] ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:627 Where there some changes? Thanks Klaus

2 2

Simple command to insert attributes to domain_attrs
by Philippe Sultan 17 Jan '13

17 Jan '13

Hi, I'm migrating from sip-router/Kamailio 3.1, SER flavour to the latest dev version, with MySQL support. I need to add attributes to domains, and therefore use the domain_attrs table. A handy command I used for that was ser_attr : ser_attr add domain=anydomain.voip attr=value Is there something similar I can use under the Kamailio flavour? kamctl does not seem to help here, unless I missed an option. Regards, Philippe

3 5

Users can't authenticate via external IP
by Rumen Mihailov 17 Jan '13

17 Jan '13

Hello Kamailio guys, I have the following setup Internet -----> REAL STATIC IP -----> Kamailio Virtual IP(10.2.45.101) I have setup everything and when I try to connect to kamailio from the inside I can successfully register. When I try to register to the REAL STATIC IP which is forwarded to 10.2.45.101 I get unauthorized. I have narrowed down the problem to this: auth_check("$fd", "sip", "1") as I said I can authenticate from the internal network but not from the Internet. What do I need to alter ? Side problem... why when using xlog("L_WARN","$Cxx TEST") kamailio fails to start ? seems that my version does not support the colored output... Thank you for your help guys...in advance. Best regards, Rumen http://itradebg.com

2 1

Server sip.iptel.org did not respond to register
by Faisal Farooq 17 Jan '13

17 Jan '13

Dear Support, We are using IPTEL SIP services. Now we are getting Error message on our VOIP software is " Server sip.iptel.org did not respond to register". Moreover, we can successfully ping and complete trace route the sip.iptel.org from our server but failed to register through sip.iptel.org server. Kindly suggest what is the cause and look into this matter. I am sending you our IPTEL SIP ID/Username and please find below. (1) 196211 (2) 195404 Regards, Faisal

2 1

i cannot log on to iptel.org
by kelly randolph 17 Jan '13

17 Jan '13

i cannot log on to iptel.org is everything okay???? thanx, kellyr

2 1

cant login
by Pasindu Hettiachchi 17 Jan '13

17 Jan '13

hi , my linksys router failed to register with iptel.org this happened from last night, any think that I have to change ? Pasindu

2 1

Error while loading modules, if given a custom library path
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi I have trouble loading modules from a custom lib directory. It works fine if the libraries are located under PREFIX/lib/Kamailio/ (PREFIX is the path that I gave while building Kamailio). But on the target I will have to keep the libraries in a custom directory. I tried by changing the mpath in kamailio.cfg. Also I tried by giving the path in -L option while invoking kamailio. It is not working. It fails with the following errors, even if the libraries are present at the location where it compains. I searched but couldn't find any answer why it is failing, though it is supposed to work. Please help me. $ ./sbin/kamailio -E -d -f /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg -c loading modules under /home/ajay/projects/video_conf/lib/kamailio/:/home/ajay/projects/video_conf/lib/kamailio/modules/:/home/ajay/projects/video_conf/lib/kamailio/modules_k/ 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 222, column 12-23: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/kex.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 223, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/tmx.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 225, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules/sl.so>: libkcore.so.1: cannot open shared object file: No such file or directory $ ls -l /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so -rwxr-xr-x 1 ajay ajay 155651 Jan 16 09:58 /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so $ ls -l /home/ajay/projects/video_conf/lib/kamailio/libkmi.so* lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so -> libkmi.so.1.0 lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1 -> libkmi.so.1.0 -rwxr-xr-x 1 ajay ajay 78111 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1.0 Regards, Ajay

1 1

make option to build a lean kamailio image
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi, I need to build a simple Kamailio server that could do just registration and basic call routing. I don't need database support, it is fine if it loses data during reboot. The only other option I may need is NAT support. So far I was building the server using make all as given below: $ make FLAVOUR=kamailio PREFIX="$HOME/projects/video_conf/sipserver" cfg $make all $make install I noticed that the total size of library and binary files is huge, little less than 30MB. I have to port and test the server on a target where I don't have much space. Can somebody please suggest me what make option would be appropriate for me? Is it fine to not build the ser modules under modules_s/ and what is make option to do so? I tried "make" instead of "make all", but it somehow builds everything during the step "make install" and the final result is same. Thanks, Ajay

2 2

TLS questions
by Fabian Borot 15 Jan '13

15 Jan '13

Hi, we need to implement TLS in our current setup and have some questions about it: 1- We have a farm of Asterisk servers behind our kamailio proxy. The call flow is Customers/Providers <--> Proxy <--> Asterisks. Is it possible to have TLS only between the Customers/Providers and the Proxy and then just regular SIP over UDP between the Proxy and the Asterisks? 2- Can we have TLS over both TCP and UDP at the same time? Thank you a lot fborot

3 3

Dokuwiki upgraded
by Daniel-Constantin Mierla 15 Jan '13

15 Jan '13

Hello, fyi, I upgraded dokuwiki to latest version, switching to the new default template (which is not bad at all, btw) as the old one is not yet fully compatible. If you find any issue, report to devel mailing list. It is about the wiki for v3.2+: - http://www.kamailio.org/wiki/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

NAPTR for TLS only
by Daniel Pocock 14 Jan '13

14 Jan '13

I'm just wondering if anyone can comment on expected and actual behavior if there is only a NAPTR record for TLS, e.g. I have: sip5060.net. IN NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.sip5060.net. and I don't have any entry for "SIP+D2U" or "SIP+D2T" If some third party Kamailio instance (e.g. sip-server.example.org) receives a request from a user trying to call sip:user@sip5060.net, with a sip: rather than sips: URI, should it (and will it) use the "SIPS+D2T" result, if no other result is available? Or would it ignore the NAPTR record and try to find the default SRV record such as _sip._udp.sip5060.net ? Should there be another NAPTR record to translate sip: to sips: using a regex perhaps, or would such a NAPTR be a bad thing?

3 4

Kamailio, Asterisk with TLS+SRTP
by Roberto Fichera 14 Jan '13

14 Jan '13

Hi All, I would setup a configuration where Kamailio authenticate asterisk SIP trunk using TLS and SRTP. At moment I was able to configure everything, including RTTProxy since most of the asterisks v1.8.19.1 are behind NAT. So far so good it works pretty good using standard authentication and the call goes straight between asterisks. But as soon as I move my configuration for both kamailio & asterisk to TLS+SRTP I'm not able to authenticate asterisk SIP trunks. Especially asterisk seems insisting to use the port 5060 even if I requested the TLS on 5061. kamailio v3.3.3 tls.cfg is configured as: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/server.key certificate = /etc/pki/tls/certs/server.pem ca_list = /etc/pki/tls/certs/ca-bundle.crt #crl = //etc/kamailio/crl.pem # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] verify_certificate = no require_certificate = no So my asterisk conf is the following: [general] tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/5002.pem tlscafile=/etc/asterisk/ca-bundle.crt tlscipher=ALL tlsclientmethod=tlsv1 tlsdontverifyserver=yes transport=tls,udp .... ..... and the SIP trunk is configured as [kamailio] type=peer insecure=invite,port nat=yes disallow=all allow=ulaw host=kamailio_ip outboundproxy=tls://kamailio_ip port=5061 defaultuser=5002 fromuser = 5002 fromdomain =mydomain secret=5002 qualify=yes dtmfmode=rfc2833 context=default callbackextension=5002 directmedia=nonat sendrpid=yes transport=tls encryption=yes register => tls://5002:5002@kamailio_ip:5061/5002 I still get error like: Jan 14 10:45:12] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:12] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:32] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:32] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:52] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:52] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:46:07] ERROR[7041]: tcptls.c:444 ast_tcptls_client_start: Unable to connect SIP socket to kamailio_ip:5060: Connection timed out Does anyone can suggest me something to read, try, check? Best regards. Roberto Fichera.

2 2

How to correctly check if the variable is set?
by Mino Haluz 14 Jan '13

14 Jan '13

Hi, how should I check if the value is set? if ($avp(s:test) == "") { or is there any null keyword ? If so, does it work for $avp, $sht, $var and $shv ? Thanks, Mino

3 4

Development is frozen for v4.0.0
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, last midnight was the term to freeze the development for next major release v4.0.0. For a while, the GIT must branch should get only commits related to bug fixes, documentation improvements and updates of the additional tools. If any developer needs to store code of new features in the repository, use personal branches (or temporary branches in tmp/ when many developers work on the same code). After unfreeze, these branches can be merged in the master branch. For 1-2 months from now on, the main focus with the master branch is testing. The v4.0.0 will be out after mid of February, exact date to be set later based on the results during testing phase. There were many additions, so any kind of help testing is very appreciated. Fortunately the core and most of the very used classic modules were not changed much, ensuring good stability. But there are old very used pieces such as usrloc/registrar that got touched as well, so try to test as much as possible the components you are using in your configs. Provide any kind of feedback on sr-dev mailing list or the bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

PV: PAI vs. PPI
by Klaus Darilion 11 Jan '13

11 Jan '13

Hi! I just found out that there is only $ai for the whole P-Asserted-Identity URI, but lots of PVs for P-Preferred-URI: uri, domainpart, userpart ... Is there a reason for that, or is it just that nobody implemented it for PAI? I think it should be easy to extend it for PAI too. regards Klaus

2 1

Kamailio World Conference - Registration is open
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, the registration for Kamailio World Conference is now open! You can see more details and register at: - http://conference.kamailio.com/k01/registration/ There is already a great group of speakers and interesting proposed talks. More regarding the content will be published in the near future, keep an eye on event's web site: - http://conference.kamailio.com If you are considering to speak at the conference, submit your proposal as soon as possible, the slots are filling up quickly. Looking forward to meeting many of you at the conference! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 3

Kamailio on Twitter
by Olle E. Johansson 11 Jan '13

11 Jan '13

Hello! For those of you that have missed it: You can now follow the project on Twitter: @kamailioproject https://twitter.com/kamailioproject Amongst the latest news are Kamailio at Fosdem 2013, Code freeze for kamailio 4.0 and the Kamailio World conference now open for registrations. Tonight we went into code freeze for the coming release of Kamailio 4.0. This means that developers can not add any more features to this release, only fix bugs. In order to fix bugs we will need your help testing. After the weekend we'll come back with instructions on how to test and report bugs. The information is already available on the web site. Have a great weekend! /O

1 0

LCR
by Douglas Ugalde 11 Jan '13

11 Jan '13

Hi, Im trying to configure LCR in Kamailio 3.3.3 but I dont Know how can I do to fix this error: ERROR: lcr [lcr_mod.c:1840]: error while parsing R-URI This is my LCR configuration block: route[LCR] { if(!load_gws(1)){ sl_send_reply("500", "Internal server error, unable to load gateways"); xlog("L_NOTICE","Internal server error, unable to load gateways"); break; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); break; } } I not sure if this configuration thats ok, please somebody help me. Note: Params and modules are already loaded, sorry for my english. Best regards.

2 1

Problem with MSILO module forwarding messages
by Owen Lynch 10 Jan '13

10 Jan '13

Hi, I have the kamailio 3.3.2 msilo module basically working but I cannot configure it to forward stored messages to the registered UA. Instead it forwards to itself (via DNS lookup on the domain name). This is causing me grief because I am getting recursive message handling. The module overview describes this behaviour: "Every time when a user registers with Kamailio, the module is looking in database for offline messages intended for that user. All of them will be sent to contact address provided in REGISTER request." The REGISTER request contact header looks OK: Contact: <sip:hxqecvtn@192.168.2.236:57532 ;transport=tls>;expires=3600;received="sip:192.168.2.236:53628 ;transport=TLS";+sip.instance="<urn:uuid:1ee63413-3569-46dc-a9ee-34c6e4e1403c>" However the 'outbound_proxy' parameter seems to contradict this: "The SIP address used as next hop when sending the message. Very useful when using Kamailio with a domain name not in DNS, or when using a separate Kamailio instance for msilo processing. If not set, the message will be sent to the address in destination URI." So, when 'outbound_proxy' is not set, the message is sent to the address in the destination URI, rather than the UA contact address? Can the module be configured to forward stored messages directly to the UA contact address? Thanks, Owen Lynch

2 1

Socket error
by Douglas Ugalde 10 Jan '13

10 Jan '13

Hi, Please I need help with this error: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) I'm using Kamailio 3.3.3 I call 2 times before got this error, how can I do to resolved it, I really really appreciated any help. I attached my Kamailio.cfg. Really sorry for my English. Best Regards.

2 1

NAT Problem! Not relay ACK,CANCEL and BYE to other UA
by Nguyen Anh Tuan 10 Jan '13

10 Jan '13

Hi All, I have 2 interfaces - External A (behind NAT) and Internal B (LAN) in Kamailio Load Balance, and enable Multi-Homing function "mhomed=1". Load Balance dispatches to Kamailio SIP Servers via the Internal B (LAN) (each Kamailio SIP server has both proxy and registrar in one box). For RTP relay, I use RTPproxy in other server, has 2 interfaces A&B like Load Balance. If using 2 interfaces in Load Balance, Registration and calling NORMAL with 2 users behind NAT, but the call will be interrupted after 30 seconds due to the ACK message (after Ringing, 200 OK) STOP in Load Balance in return and not relayed to other UA . CANCEL and BYE have same error like ACK. I tried using only one interface for load balancing (External = Internal IP), and SIP Server binds to Load Balance in the same network, EVERYTHING IS OK. ACK, CANCEL and BYE are not missing behind NAT. For NAT Traversal I use PATH Module in Load Balance, and add "Received" like this: ... modparam("path", "use_received", 1) .. .. record_route(); add_path_received(); force_rport(); t_relay; .. .. And in SIP Server enable Path Parameter for Registrar: modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) Enable "#!define WITH_NAT" in default kamailio.cfg for RTP Controlling, but I removed all fix_nated_register and fix_nated_contact in SIP Server, because using PATH module.I try enable both fix_nat_register and fix_nated_contact, or only fix_nat_register but NOT SOLVE Problem "missing ACK,CANCEL, and BYE in Load Balance" behind NAT. Best Regard, Tuan.

2 1

Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
by Richard Brady 10 Jan '13

10 Jan '13

Hi Peter Great work on this! We'd like to help you test. The test I have in mind, which we could create using SIPp, would be to register multiple contacts with the same instance-id (i.e. "sip.instance" param) but different reg-id params. Then send an INVITE to that AoR and make sure the forking is only per instance-id and not per reg-id. This could be repeated in multiple permutations of instance-ids and reg-ids. This would be a test of save() and lookup() more than anything else. Is that what you had in mind? Richard On 3 January 2013 14:13, Peter Dunkley <peter.dunkley(a)crocodile-rcs.com>wrote: > I hope to get the outbound edge proxy and flow timer stuff into master by > Monday, but it could really do with some additional testing especially in > conjunction with the reg-id stuff in registrar/usrloc (which I have no idea > how to use). > > Peter > >

5 14

LCR with Siremis
by Douglas Ugalde 09 Jan '13

09 Jan '13

Hi, I was configuring the modparams for LCR and Dialplan, I taken from kamailio web site, but I think the params are outdated, because we need to erase some lines to match with the tables in the DB, Im using Kamailio 3.3.3 and Siremis 3.3 Please can you give 1 example about how to make outbounds calls with siremis 3.3 (send calls to gateway)? I already have the lcr and dialplan module loaded with params. Where can I get update documentation about Siremis 3.3 and Kamailio 3.3.3. Thanks for your help.

1 0

Freezing development for v4.0.0
by Daniel-Constantin Mierla 09 Jan '13

09 Jan '13

Hello, a (26-hours before) reminder that development for next major release v4.0.0 will be frozen tomorrow, Thursday, January 10, 2013, at 23.59 GMT. If any developer has new features to push in the git master branch, please do it before. After that, for a while, the master should get only bug fixes and improvements to documentation and addition tools (e.g., bringing kamctl up to date). New features have to be kept in personal branches until the master branch is open again. The focus should be on testing from the day after tomorrow on, I hope the users community will help us here as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Having "modparam" options in the database
by Glenn O. Larsen 09 Jan '13

09 Jan '13

On Tue, Jan 8, 2013 at 9:21 AM, Olle E. Johansson <oej at edvina.net> wrote: >>>> Is there a way to read modparam config from a database? > In most cases, there should be no need to change modparams at > runtime. > > Can you tell us more about which parameter you want to change during > runtime that doesn't have a mI/rpc reload command? Hi Olle! The module we have been using, and wanted database configuration for was the pike module. But and after spending some more time looking at alternative modules, I see that we can use other modules to help us with traffic control/blocking, like the pipelimit and the ratelimit (both these modules have some sort of database integration, need to look more into this). I was so focused on the pike module, that I didn't see these other modules.. :) -- mvh Glenn O Larsen

3 3

Core fm_realloc function
by Bruno Bresciani 08 Jan '13

08 Jan '13

Hi all, I observed two at 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536. I don't know exactly what may be caused this core because the debug of kamailio is disabled, but looking at backtrace (gdb) and source code I could notice a relationship with TLS connection establishment (handshake). Other important fact is that core was generated by the kamalio itself with signal 6 (abort) because it can't find the pointer to memory realloc. I'd like to know if someone already observed this issue and if my analyse above is correct? Bellow is backtrace full with gdb: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00f30402 in __kernel_vsyscall () (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 0x00f30402 in __kernel_vsyscall () No symbol table info available. #1 0x009fec10 in raise () from /lib/libc.so.6 No symbol table info available. #2 0x00a00521 in abort () from /lib/libc.so.6 No symbol table info available. #3 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536 f = (struct fm_frag *) 0xb61f0bb8 pf = <value optimized out> orig_size = 18 L=ÃbÂ¬Ã2n = (struct fm_frag *) 0xb61f0bd2 ptr = <value optimized out> hash = 23548 #4 0x0051a87f in ser_realloc (ptr=0xb61f0bc0, size=32) at ../../mem/shm_mem.h:266 No locals. #5 0x00460295 in OPENSSL_cleanse () from /lib/libcrypto.so.6 No symbol table info available. #6 0x004609f7 in CRYPTO_realloc () from /lib/libcrypto.so.6 No symbol table info available. #7 0x003efb22 in sk_insert () from /lib/libcrypto.so.6 No symbol table info available. #8 0x003efbaa in sk_push () from /lib/libcrypto.so.6 No symbol table info available. #9 0x007c2ac3 in ssl_bytes_to_cipher_list () from /lib/libssl.so.6 No symbol table info available. #10 0x007aa9f3 in ssl3_get_client_hello () from /lib/libssl.so.6 No symbol table info available. #11 0x007ab815 in ssl3_accept () from /lib/libssl.so.6 No symbol table info available. #12 0x007c1c3a in SSL_accept () from /lib/libssl.so.6 No symbol table info available. #13 0x005278e3 in tls_accept (c=0xb621af88, error=0xbf91dfc8) at tls_server.c:346 ret = <value optimized out> ssl = (SSL *) 0xb61c1878 cert = <value optimized out> tls_c = (struct tls_extra_data *) 0xb6208e58 tls_log = <value optimized out> #14 0x0052da8e in tls_read_f (c=0xb621af88, flags=0xbf91e200) at tls_server.c:1028 r = (struct tcp_req *) 0xb621affc bytes_free = 4095 bytes_read = 70 read_size = 4095 ssl_error = 0 ssl_read = 0 ssl = (SSL *) 0xb61c1878 rd_buf = "\026\003\001\000A\001\000\000=\003\001OÃ¿\vÂ¡ywP\fkÃDÃ 0/\000\a\000\005\001\000ÃÃÃÃ¯000\000\026\0009\0008\0005\000\026\000\023\000\n\0003\0002\00 202i<Ã¤<Ã¥Ã¾\031Ã¬234Â \237Ãª030Y){lÑ¶"...06Ã'Ãr9Ã¬Ã«Â¸{[Â©\t\205\037\036Ã¿HPH5Âµ\t(Ã©32Ã027Ã¦Ã²036Â¾\"\rÔ¤YÃ215uÂº\027)Ã¶000Ã±215\vÃÃ Ã©Ea\023xÃ©J\206T\204CÃ wr_buf = "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"... rd = {buf = 0xbf90dfa4 "\026\003\001", pos = 70, used = 70, size = 65536} wr = { buf = 0xbf8fdfa4 "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"..., pos = 0, used = 0, size = 65536} tls_c = (struct tls_extra_data *) 0xb6208e58 enc_rd_buf = (struct tls_rd_buf *) 0x0 n = 0 flush_flags = <value optimized out> err_src = 0x541a83 "TLS read:" x = <value optimized out> #15 0x08158ade in tcp_read_headers (c=0xb621af88, read_flags=0xbf91e200) at tcp_read.c:406 bytes = -1 remaining = <value optimized out> p = <value optimized out> r = (struct tcp_req *) 0xb621affc #16 0x08158fd4 in tcp_read_req (con=0xb621af88, bytes_read=0xbf91e204, read_flags=0xbf91e200) at tcp_read.c:871 bytes = <value optimized out> total_bytes = 0 resp = <value optimized out> size = <value optimized out> req = (struct tcp_req *) 0xb621affc dst = {send_sock = 0x1, to = {s = {sa_family = 57540, sa_data = "\221Â¿\020\000\000\000\000\000\000\000\020\000\000"}, sin = {sin_family = 57540, sin_port = 49041, sin_addr = {s_addr = 16}, sin_zero = "\000\000\000\000\020\000\000"}, sin6 = {sin6_family = 57540, sin6_port = 49041, sin6_flowinfo = 16, sin6_addr = {in6_u = {u6_addr8 = "\000\000\000\000\020\000\000\000\001\000\000\000\001\000\000", u6_addr16 = {0, 0, 16, 0, 1, 0, 1, 0}, u6_addr32 = {0, 16, 1, 1}}}, sin6_scope_id = 12}}, id = -1080958468, proto = 4 '\004', send_flags = {f = 0 '\0', blst_imask = 0 '\0'}} c = 10 '\n' #17 0x08159a0f in handle_io (fm=0x83296b0, events=1, idx=-1) at tcp_read.c:1150 ret = 4 n = 4 read_flags = 1 con = (struct tcp_connection *) 0xb621af88 s = 12 resp = <value optimized out> t = <value optimized out> #18 0x0815d653 in tcp_receive_loop (unix_sock=35) at io_wait.h:1091 No locals. ---Type <return> to continue, or q <return> to quit--- #19 0x0812885b in tcp_init_children () at tcp_main.c:4819 r = 3 reader_fd_1 = 35 pid = <value optimized out> si = <value optimized out> #20 0x080ad412 in main_loop () at main.c:1632 i = 4 pid = <value optimized out> si = (struct socket_info *) 0x0 si_desc = "udp receiver child=3 sock=192.168.166.31:5060 \000\221Â¿\002\000\000\000/\200\220\000\001\000\000\000\000\000H\000Â¨Ã¤21Â¿\001\000\000\000Â°Â¶Ã¨026Â¶\000\000\000\000\000ÃÂµ\030s2\b\002\000\000\000Ã¢f\b\000ÃÂµ\001\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000Ã¿Ã¿Ã¿Ã¿Â¸Ã¤21Â¿" #21 0x080afbe7 in main (argc=3, argv=dwarf2_read_address: Corrupted DWARF expression. ) at main.c:2398 cfg_stream = (FILE *) 0x8a2d008 c = <value optimized out> r = 0 tmp = 0xbf91e674 "\212Ã¾\221Â¿\233Ã¾\221Â¿Â°Ã¾\221Â¿Â»Ã¾\221Â¿Ã\221Â¿Ã¹1Â¿\020Ã¿\221Â¿DÃ¿\221Â¿LÃ¿\221Â¿WÃ¿\221Â¿]Ã¿\221Â¿oÃ¿\221Â¿{Ã¿\221Â¿\202Ã¿\221 tmp_len = 10492229 port = 136192361 proto = -1080957480 ret = <value optimized out> seed = 21782655 rfd = 4 debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = <value optimized out> p = <value optimized out> Best Regards

4 24

ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES)
by Shreyansh Purwar 08 Jan '13

08 Jan '13

Hello Everyone, While creating an account on Kamailio Sip server (I am running the command "kamctl add test test1"), it is asking for password of MySQL Server and after entering the password, I am getting following errors :- - ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES) - ERROR: introducing the new user 'test' to the database failed Please help me in resolving this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

raw_query is crashing....
by Krishna Kurapati 08 Jan '13

08 Jan '13

Hi, Since there is no way to limit the result set to 1, I tried to use raw_query inside msilo module. Everytime it executes, it is crashing. The same query if I run in mysql shell, it is returning successfully. Here is the code That I added: query_len = snprintf(query, 2048, "SELECT * FROM %.*s WHERE %.*s='%.*s' AND %.*s='%.*s' AND %.*s!=%i AND %.*s<%i LIMIT 1;", ms_db_table.len, ms_db_table.s, sc_uri_touser.len, sc_uri_touser.s, puri.user.len, puri.user.s, sc_uri_tohost.len, sc_uri_tohost.s, puri.host.len, puri.host.s, sc_status.len, sc_status.s, 200, sc_status.len, sc_status.s, MSG_NOTIFY_OFFSET); if (query_len < 0) { LM_ERR("error in sql snprintf"); goto error; } LM_DBG("m_dump_msg: Query=%s\n", query); query_str.s = query; query_str.len = query_len; if (msilo_dbf.raw_query(db_con, &query_str, &db_res) < 0) { LM_ERR("Failed to query database. Query = %s\n", query); goto error; } The Debug Log at Level 4 shows following: 11(7520) DEBUG: msilo [msilo.c:1120]: m_dump_msg: Query=SELECT * FROM silo WHERE tousername='968257197' AND todomain='50.62.1.7' AND status!=200 AND status<1000 LIMIT 1; 11(7520) DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x7fdd4b605540 11(7520) DEBUG: db_mysql [km_res.c:68]: 16 columns returned from the query 11(7520) DEBUG: <core> [db_res.c:155]: allocate 128 bytes for result names at 0x7fdd4b5d8dc0 11(7520) DEBUG: <core> [db_res.c:165]: allocate 64 bytes for result types at 0x7fdd4b5d8e60 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x7fdd4b5d8d90 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d90)[0]=[id] 11(7520) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x7fdd4b5d8d60 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d60)[1]=[src_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x7fdd4b5d8d30 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d30)[2]=[dst_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type <skip> 11(7520) DEBUG: <core> [db_val.c:117]: converting STRING [ac3aeddd9b6f45e6822c824aa2c133b2] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [0] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [1] 26(7552) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 22 26(7552) DEBUG: <core> [tcp_main.c:3592]: DBG: handle_ser_child: dead child 11, pid 7520 (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 22, -1, 0x0) fd_no=44 called 26(7552) DEBUG: <core> [tcp_main.c:3353]: DBG: handle_tcp_child: dead tcp child 1 (pid 7520, no 11) (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 25, -1, 0x0) fd_no=43 called 0(7509) ALERT: <core> [main.c:787]: child process 7520 exited by a signal 11 0(7509) ALERT: <core> [main.c:790]: core was not generated 0(7509) INFO: <core> [main.c:802]: INFO: terminating due to SIGCHLD As you can see something happening to Database connection. Not sure why I am unable to use raw_query. Normal API query from msilo works just fine. What am I doing wrong to make it crash? Thanks Krish Kura

2 1

Having "modparam" options in the database
by Glenn O Larsen 08 Jan '13

08 Jan '13

Hi guys, I have tried to Google this for some time now, but can't seem to find an answer. Is there a way to read modparam config from a database? -- Best regards, Glenn Larsen

3 3

LCR weight
by Ricardo Martinez 08 Jan '13

08 Jan '13

Hello. I have a doubt about the WEIGHT value in the LCR module. In the past we used the Kamailio version 1.5.2 and in that version LCR weight was calculated by a little script called lcr_weight_script.php. There was not “percent” values like 50% and 50% for two gateways despite that the probabilities were numbers calculated by the script. My question is : For Kamailio version 3.2.4 do I have to use the same script to calculate the weight?? I was almost sure that the weight column was a probability value. For example if I have 3 gateways, with 50% 25% and 25% The weight column should be : 50, 25 and 25 ? or 45, 32 and 32 as the script indicates??? [root@vlcr utils]# ./lcr_weight_test.sh 45 32 32 weight 45 probability 0.5235 weight 32 probability 0.2352 weight 32 probability 0.2413 Please enlighten me here. Thanks Ricardo.-

2 1

TCP Path received quotes
by Spencer Thomason 07 Jan '13

07 Jan '13

Hello, I'm forwarding registrations to Freeswitch and adding a Path header using add_path_received() for endpoints behind NAT. When TCP is used, this results in a header like this: Path: <sip:a.a.a.a:5070;lr;received="sip:b.b.b.b:35430;transport=tcp"> Freeswitch doesn't like the transport protocol inside the quotations and refuses to route a call to the registered user. Constructing the header manually without the quotes such as: append_hf("Path: <sip:$Ri:$Rp;lr;received=sip:$si:$sp;transport=$pr>\r\n"); solves the problem. Is there a reason for the quotes? Best Regards, Spencer

4 7

Multi tenant Kamailio
by Rumen Mihailov 07 Jan '13

07 Jan '13

Hello Kamailio, VoIP and asterisk gurus, I have the following really complicated scenario to solve. I have CompanyA which has extensions 100-120 I have CompanyB which has extensions 100-115 I have CompanyC which has extensions 100-130 Now I want to have all those extensions on a single kamailio box, but I want to achieve the following. CompanyA calls 110 and this rings CompanyA 110 not CompanyB or CompanyC. What I want to achive is multitenancy. I know how to make this in asterisk but I want to take advantage of Kamailio's load balancing and HA features. Sammy from the mailing lists advised that I use multidomain and actually I saw that this module is indeed intended to do that, however I have no clue how to use it. I mean do I have to setup a domain for all the companies ? For instance companya.itradebg.com, companyb.itradebg.com and then advise the customers to register to their respective domain or what ? This concept is really a bit tough for me despite all the help from Sammy for which I am really greatful. I would greatly appreciate any tutorials or links to some useful documentation apart from the module documentation. Thank you in advance guys. Best regards, Rumen http://itradebg.com

2 2

NAT'ed RLS setup
by Dmytro Bogovych 07 Jan '13

07 Jan '13

Greetings. I have kamailio 3.3.3 setup on server behind the nat. Nat forwards ports for tls/tcp/udp. This setup worked without rls ok. I added necessary modules - rls, pua, xcap etc. Updated a config file and db. Client application uploads xcap documents ok; after app sends subscribe request to list address. Rls/pua modules generate subscriptions to entries in list. But dns resolver finds the public ip as target for subscription. The result is: sip packets cannot reach the server - they are sent to public ip of nat machine. I see it in ngrep output. Nat does not route them back. I add the domain name and _sip._udp.domain with internal ip address to /etc/hosts. But kamailio does not use /etc/hosts when dns caching is active. I do not want to loose dns caching. I made an attempt to run without caching but server was not stable. It crashed couple of times - so i stopped these attempts. Is there way to put eternal domain -> ip bindings into dns cache? I want to put bingings to internal ip - hope it will help. Maybe there is another way to handle this problem?

1 0

Timeout for t_uac_dlg per request
by David 07 Jan '13

07 Jan '13

Hello, I am running Kamailio 3.2.x, I am using udp mi to send a t_uac_dlg command, :t_uac_dlg: NOTIFY sip:phone1@proxy.lan . . "From:sip:daemon@proxy.lan To:<sip:phone1@proxy.lan> Contact:sip:daemon@proxy.lan" # I would like to know if I can change the timeout for this request. Running a packet sniffer on the proxy, I see the packet is retransmitted after 500ms, then 1000ms, then 2000ms, etc... This is the proper behaviour for SIP, so we can see my tm module is working well. My problem is that I want a response within 3 seconds ( I know this doesn't respect the normal SIP timer values ). I do not want to change the timers in the tm module for the entire proxy, so my question is if there is a way to set the timeout for the dialog created by the t_uac_dlg command, and can I set it on a per dialog basis ? David

1 0

TLS with self-singned certificates
by Iurii Andamasov 07 Jan '13

07 Jan '13

Hello, trying setup kamailio with TLS, have 2 peers, tls.cfg: http://pastebin.com/DvdDzx0v i'm getting Jan 4 10:38:43 fs-tls /usr/sbin/kamailio[3798]: ERROR: tls [tls_server.c:1190]: TLS read:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure Can someone point me to solution? Thanks

3 4

tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory
by Shreyansh Purwar 07 Jan '13

07 Jan '13

Hello Everyone, While building kamailio server 3.0.0 with TLS(I have removed tls from exclude module variable) I am getting following error :- - tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory Please help me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Malicious traffic comming from 37.75.210.90
by Nick Khamis 06 Jan '13

06 Jan '13

Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N.

1 0

dialog ka_timer and ka_interval problem
by Rinor Hoxha 06 Jan '13

06 Jan '13

Hi, I had an issue with dialog module of kamailio v3.3.3. Is the same described on http://lists.sip-router.org/pipermail/sr-users/2012-November/075735.html. I think the issue is related to dgl_hash.c on line:192 *dlg_ka_list_head = NULL; should be *dlg_ka_list_tail = NULL; and line:216 *dlg_ka_list_tail = dka; should be *dlg_ka_list_head = dka;. I will test it as soon as I can, but meanwhile I'm attaching the patch that should fix it so you can test it and report back the results if you want to. Br, Rinor

2 1

Kamailio DB version error
by Douglas Ugalde 06 Jan '13

06 Jan '13

Hi, I want to load the "dialplan.so" with parameters but when I reload kamailio, I got this error: /usr/local/sbin/kamailio[1348]: ERROR: <core> [db.c:439]: invalid version 0 for table my_table found, expected 1 (check table structure and table "version") Im using kamailio 3.3.3 Please what can I do to resolve it? Regards.

2 1

Next major release proposed as v4.0.0
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, using v4.0.0 for the next major release was proposed and debated at previous development IRC meeting. First was about the addition of a new transport layer (websockets) along with other important new features. During the debate was touched the completion of kamailio-ser integration (no more duplicated modules) as another milestone for increasing the first digit (which was finished just before the end of last year). Now we also have the IMS modules, pushed in the GIT master branch few days ago. It seems that there are lot of good reasons for numbering the next major version 4.0.0. Other comments, opinions? Have a great and happy year ahead, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 4

Multiple rtimer processes + mqueue
by Alex Balashov 04 Jan '13

04 Jan '13

Will multiple rtimer processes consuming the same mqueue, i.e. via a blocking mechanism while(mq_fetch(...)) ... hurt anything? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Freezing for next major release
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, based on last irc devel meeting and feedback afterwards, most of the people seem to be fine releasing next major release a bit earlier, in order to: - make websockets support and the other new features available in a stable version - allow proper development time frame for other planned new features that wouldn't have been ready if the release would be 1-2 months later So, considering the winter holidays, I propose to freeze development on January 7, 2013, with full release about one month later. It seems that many people are already using devel version to play with websockets, meaning that we are staying very well with testing. Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 5

"maximum number of branches exceeded" after kamalilio restart
by Andrew Pogrebennyk 04 Jan '13

04 Jan '13

Hello, I'm getting some errors - if I restart kamailio on a live system, the first call after restart fails with "max nr of branches exceeded". However, we are not doing append_branch() or anything fancy for that call at all: Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: <core> [dset.c:306]: max nr of branches exceeded Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: registrar [lookup.c:298]: failed to append a branch Also it appears on the other system with a different config: Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:656]: ERROR: add_uac: maximum number of branches exceeded Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed The second call however goes through and this error does not appear until the next kamailio restart. What could be the problem here? P.S. The kamailio version is 3.3.2. Andrew

1 0

km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory
by Shreyansh Purwar 04 Jan '13

04 Jan '13

Hello Everyone, I need a help, I was installing kamailio SIP server according to the instruction provided at wiki page - http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git but while compiling it(make all) i am getting following error - km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory Can anybody tell me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Re: [SR-Users] [sr-dev] Freezing for next major release
by Peter Dunkley 03 Jan '13

03 Jan '13

I hope to get the outbound edge proxy and flow timer stuff into master by Monday, but it could really do with some additional testing especially in conjunction with the reg-id stuff in registrar/usrloc (which I have no idea how to use). Peter Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >Hello, > >a reminder that the freezing date of the development for the next major >release is next Monday, January 7. > >Considering it was vacation time so far, we can prolong development for >few days if it is needed - just reply here. > >Besides few small additions, I want to add internal hash table to track >MSRP connections, hopefully I should be able to do it in time. If not, I >will ask for few more days as well. > >Cheers, >Daniel > >On 12/5/12 5:05 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> based on last irc devel meeting and feedback afterwards, most of the >> people seem to be fine releasing next major release a bit earlier, in >> order to: >> - make websockets support and the other new features available in a >> stable version >> - allow proper development time frame for other planned new features >> that wouldn't have been ready if the release would be 1-2 months later >> >> So, considering the winter holidays, I propose to freeze development >> on January 7, 2013, with full release about one month later. >> >> It seems that many people are already using devel version to play with >> websockets, meaning that we are staying very well with testing. >> >> Other opinions? >> >> Cheers, >> Daniel >> > >-- >Daniel-Constantin Mierla - http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > >_______________________________________________ >sr-dev mailing list >sr-dev(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

1 0

recreating kamilio_fifo without restart
by Uri Shacked 03 Jan '13

03 Jan '13

Hi, I entered the command "kamailio -v" instead of "kamailio -V". It looks like it starts kamailio again even that it was running.... the problem is that the kamctl_fifo and kamctl_ctl file are gone now. how do i create the files again without restarting kamailio? Thanks, Uri

3 4

All The Best And A Happy 2013!
by Daniel-Constantin Mierla 31 Dec '12

31 Dec '12

This was it for 2012, one of the greatest years in the history of the project! One major release was done in the summer (version 3.3.x), another one is few days before testing phase, most probably to be out by mid of February. The next major release includes several great features and edevelopments: - support for websockets - IMS extensions in the main GIT branch - no more duplicated modules, integration of Kamailio and SER ended There are big plans for 2013. First is the next major release, together with participation at FOSDEM conference in Berlin. A bunch of other events will open the path to our first dedicated conference for Kamailio project - Kamailio World. We hope to meet many of you at Kamailio World conference, in Berlin, Germany, during April 16-17. Last organizing bits were sorted out and registration will open in the first days of 2013 -- you can see more details at: - http://conference.kamailio.com Looking forward to a great 2013 for Kamailio and everyone involved in the project, from developers to community members! All the best! Happy New Year! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio TLS setup and certificate
by Burak ÖZTÜRK 31 Dec '12

31 Dec '12

Hi, I m a new kamailio user. Kamailio version is 3.3. I want to enable kamailio's tls support. I fallowed instructions from this link : http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates And I also enabled tls module in the kamailio.cfg file. (definig WITH_TLS and adding listen=tls:myIP:5061 command) But after trying to start kamailio i got "failed to start" error. When I checked the logs there is nothing about tls or ssl. Actually I got an error while fallowing the link above. "openssl s_client -connect localhost:5061 -tls1" command returns connection failed 111 error. I thought that this is the reason failed to start. But where or what am I doing wrong? I searched the web and hear about SRV records. But there is no explanations about this, on your guide/wiki. I also read this link too but cant get my answer : http://sip-router.org/docbook/sip-router/branch/master/modules/tls/tls.html Is there any idea or a better well-explained user guide about tls support? Regards, Burak

2 1

Zero left padding IP in Kamailio
by Techie Sup 30 Dec '12

30 Dec '12

Hello, My IP is 192.168.1.25 and I have added left zero padding i,e: 192.168.001.025 in the configuration file and try to start kamailio, it works.. but along with that rtpproxy does not start. Once I revert back it works. My script requires to add zero left padding for a specific purpose and for that I cannot remove the left zero padding. Any solution to get this work? Regards Selvam

2 1

kamailio-ser merging finished
by Daniel-Constantin Mierla 29 Dec '12

29 Dec '12

Hello, in November 2008 was decided to start integration of kamailio and ser project, to get to same source code. About one year later, there was a single source code tree, but there were several duplicated modules that kept some confusion around the two projects. As the time passed, it was shown better what modules are in the focus, making the decisions to sort them out easier. The past weeks were dedicated to removing duplicated, process that just ended -- at this moment there is no duplicated module, all of them being integrated. In summary, the modules from ser that had no conflict were moved as is to modules folder, eventually after rename. The modules with conflicts were merged, adding missing features to the most developed ones, moving the others to obsolete folder. It happened that kamailio version of the modules stayed in the majority of the cases, because they had larger set of features (the core in v3.0.0 was kept from ser). Please note that the extra features from ser modules might not be in the same form, but alternatives that should offer same functionality at the end. People using ser flavour so far should start testing the master branch. Do ask questions on the mailing lists if something that you used to have in config is not longer available, so we can assist in migration and, hopefully, create a migration guide on wiki. There might few bits and pieces to tune here and there, soon we will get into code freezing period for testing the next major release, a good time for such adjustments. It is important that everyone using former ser modules to report the issues that are found. Many thanks to all those involved in integration efforts as well as in new developments that kept the project flying! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

GROUP - change database table
by Daniel W. Graham 28 Dec '12

28 Dec '12

I am attempting to move the location of the GROUP 'grp' table. I created column grp in subscriber table and configured column with same settings as default table and updated column name as follows: modparam("group", "db_url", DBURL) modparam("group", "table", "subscriber") modparam("group", "use_domain", 1) As the other required column names are present in the subscriber table and also default params for this module, I didn't specify them, but I have also tried doing that with same result. I have debugging enabled and when attempting to restart / start kamailio it simply doesn't start and no debug errors are detected, also nothing in log file. If I change to modparam("group", "table", "grp"), kamailio starts fine. Any ideas out there as to what would cause this? Thanks for your time- -dan [cid:image001.gif@01CDDEC4.BE0AD670]

2 4

Is somebody at 29C3 in Hamburg and wants to meet for a drink and some VoIP gossip?
by Klaus Darilion 28 Dec '12

28 Dec '12

regards Klaus

1 0

No Conect
by Matheus Perillo - Rede Itatiaia 27 Dec '12

27 Dec '12

Good afternoon , My name is Matthew and Perillo on IPtel have an account with the username of djperillo . I am not able to connect with anyone. We have to work in the company of various equipment and all with Comrex IPtel account and you can not connect with any of them . Everything indicates to the page Comrex was made and does not connect . The Access are with the SIP Status On LINE . On my iphone the phone symbol on Media5 is green but when I try to connect with the Comrex Access the following message appears " user / password denied , please check your settings and try again . " What can I do to solve this problem ? Thank you . Matheus Perillo Técnico de Manutenção Tel +55 31 2105.3545 Fax +55 31 2105.3677 matheusperillo(a)itatiaia.com.br www.itatiaia.com.br Rua Itatiaia 117 • Bonfim • Belo Horizonte • MG • Brasil • 31210-170

1 0

retransmitted INVITE's
by schwatzi＠web.de 27 Dec '12

27 Dec '12

2 1

extension of SNMPStats module
by 马天 27 Dec '12

27 Dec '12

Hi all, Merry X'mas. I'm working on a project which uses kamailio as sip server. Since I'm really new to kamailio and sip protocool and c, I need some help.My question is that is it possible to extend snmpstats module so the snmpstats module can respond to customized oid.I already wrote the mib file and generated the corresponding c files which handle the request. But I couldn't compile it into snmpstats module.Always get a "No Such Object available on this agent at this OID" when i send GET request using snmpget.Thanks. Regards, Sebastian

2 1

OPTIONS packet from dispatcher module
by Morten Isaksen 27 Dec '12

27 Dec '12

Hi, I have an issue with the OPTIONS packets from the dispatcher module does not contain a max-forward header. This causes the gateway in the other end to not reply to the OPTIONS packet. Can anyone please give a hint where en the code I need to change that. I noticed that dispatcher module uses the tm module to send the OPTIONS but after that I got lost in the code. -- Morten Isaksen

2 1

Problem with RLS
by Dmytro Bogovych 26 Dec '12

26 Dec '12

Hi. I still try to setup RLS capable kamailio instance. Now the server accepts my services and resource list documents. It generated backend subscriptions - i can see it in pua table. When another peer publish its presence - presence tries to send notify to sip address specified in modparam("rls", "server_address", " sip:rls@sip.voipobjects.com"). But the notify does not come back to rls - it is rejected by somewhere in core or other module with error 404. I tried to create the rls(a)sip.voipobjects.com user - it did not help. May anyone give me advice?

1 1

Presence rules?
by Dmytro Bogovych 25 Dec '12

25 Dec '12

Greetings. I try to add rls/xcap presence into resiprocate-based user agent. May anyone tell me what module is responsible for parsing presence rules documents? I'm trying to setup "allow all" presence rules document but did not get success... Thank you!

2 4

Merry Christmas!!!
by Raj Roy Ghandhi 25 Dec '12

25 Dec '12

I WISH Merry Christmas!!! to my Kamailio FAMILY !!!! Best Regards, Roy.

1 0

obsoleting modules_s/*radius modules
by Daniel-Constantin Mierla 24 Dec '12

24 Dec '12

Hello, according to the archive, modules_s/*radius modules were kept because of duplicate and incompatible auth modules at the time there was an attempt to remove them: - http://lists.sip-router.org/pipermail/sr-users/2010-June/064318.html Now there is a single auth modules, thus I guess the modules_s/*radius can be moved to obsolete. If anyone has different opinion, reply with the arguments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 15

Queries
by Ankit sharma 24 Dec '12

24 Dec '12

Hi, I have newly installed Siremis sip router. But I have no idea How I could start or How do I start calls? If you can provide me any Manual so it would be great. Sincerely Ankit

4 3

How to set variable externally
by Mino Haluz 22 Dec '12

22 Dec '12

Hi, I would like to set my custom different debug levels (with flag?) externally with kamctl command. So I neednt restart kamailio if I want to enable/disable debug. Which module should I use in that case? Thanks, Mino

6 12

AUTOREPLY Re: Problemdebian squeze uac module load failed
by strobel＠hfo-telecom.de 21 Dec '12

21 Dec '12

******************* AUTOREPLY VON miconda(a)gmail.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 27.12.2012 bis einschließlich 01.01.2013 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Brunner oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hello, do you get another error message before? If not, set debug=3 in kamailio.cfg and run again. Then look at the logs to see if there is another message giving hints about what is wrong. Probably some mis-configured parameter. Cheers, Daniel On 12/21/12 4:38 PM, Thomas Hoellriegel wrote: > Hi all, i using debian squeeze. > I installled kamailio from the package system. > I set in the kamailio.cfg file: > loadmodule "uac.so" > I become the error: > Dec 21 16:37:35 1a-7567 > /usr/sbin/kamailio[3015]: ERROR: <core> [sr_module.c:939]: init_mod(): > Error while initializing module uac > (/usr/lib64/kamailio/modules_k/uac.so) > > Can you help plese? > thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste für blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

RPMs for v3.3.3
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, for the records, the RPMs for v3.3.3 are now available (32b and 64b: CentOS 5 and 6, RedHat 5.0 and 6.0, OpenSUSE 11.4 and 12.1, Fedora 15 and 16). They are hosted by opensuse build service, links specific per distribution are listed at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] [sr-dev] kamailio-ser integration - db modules
by Ovidiu Sas 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:20 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> What's the difference between the db1 and db2 interface? > > It is not the case here, modules_k/ldap does not have any relation to > database interface. Any database module db_[DatabaseType] implements an API. SER was using one API and opensips/kamailio a different API (and therefore today we have version 1 and 2). All db modules in the stable release are supporting both APIs. The new db ldap module has support only for the SER API version. The API is documented in each lib/srdb[1|]/*.h header files. The ldap module from ser is completely different than ldap module from kamailio (despite the fact that both are using the same name). It's the same with the dialog module. Check the README file for each. Hope this brings a little bit of light :) -ovidiu

1 0

Re: [SR-Users] [sr-dev] kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

On 12/20/12 4:13 PM, Ovidiu Sas wrote: > On Thu, Dec 20, 2012 at 9:39 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >>>> - nathelper - some extra functionality, not sure if can be kept >>>> completely >>> Maybe Andreas can look into this, as there is a lot of work going on with >>> nathelper and the new rtpproxy anyways. >> I think Ovidiu Sas looked at it when he split the rtpproxy out in a >> dedicated module. > IIRC, ping_contact whas the extra functionality in nathelper: > http://sip-router.org/docbook/sip-router/branch/master/modules_s/nathelper/… > I don't know how widely used is this functionality. > > Maybe we should have a separate thread per module (in user mailing > list to gather more imput) and see if it's worth merging the code or > use only the k version. I re-cc-ed the thread to users in case someone has comments to it. > > Also, on a separate note, I saw the we have a few db2_[module]. > I think it would make sense for these modules to rename them into > [module]_db[1|2]. > For example: ldap - we should have both versions under modules: > - ldap_db1 > - ldap_db2 > Just a suggestion ... The type of the two ldap modules are different, modules_k/ldap is a connector to ldap server from configuration file, offering possibility to do ldap search queries from config. The former modules_s/ldap (now db2_ldap) is a DB API v2 implementation driver module, so it can be used as a replacement for db_mysql (for example) when using some modules (such as db2_ops). I prefixed with db2_ to indicate that is not implementing DB API v1. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, most of the duplicated or ser-specific modules were sorted out at this time, next is the situation with the remaining ones. A) modules that will be renamed using a prefix 'uid_', because they have a database schema using unique id instead of username and domain: - auth_db => uid_auth_db - avp_db => uid_avp_db - domain => uid_domain - gflags => uid_gflags (not documented, but this module loads so called ser global avps) - uri_db => uid_uri_db B) modules to become obsolete with some work to be done - cpl-c - seems that giving a parameter with what user to be used for loading the cpl scripts will make k and s versions compatible - maxfwd - different names for config functions - nathelper - some extra functionality, not sure if can be kept completely - textops - seems to have some extra cfg functions - pike - has a rpc command - registrar/usrloc - avps stored per contact C) modules to become obsolete, most probably without further work - permissions - different structures/database table internally, but all should be achieved using k version (data migration might be required) D) modules that needs further investigations - rr - there seems to be some feature related to avps and record-route parameters/cookies - this module could be just renamed at the end (like rrs) E) Sample modules, not for real use - they can be kept in this directory (modules_s[ample]) - print - print_lib If anyone has comments regarding these remarks, reply here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

Bug in uac_replace_from if used more than once?
by Daniel Tryba 20 Dec '12

20 Dec '12

Apparantly it should be simple to mangle the From in request: http://kamailio.org/docs/modules/stable/modules_k/uac.html#id2525538 The task at hand is use the prefix 131 to suppress callerid for this call: if($rU=~"^131[0-9]+$") { strip(3); append_hf("Privacy: id\r\n"); uac_replace_from("anonymous","sip:anonymous@anonymous.invalid:$rp"); uac_replace_to("","sip:$rU@$rd"); } Just before this replace there is an other uac_replace_from to rewrite an alphanumeric username to the default callerid for that device. If there are 2 uac_replace_from in the path, things go wrong like below. A uac_restore_from() before the second uac_replace_from doesn't change this behavior. device calls 1310123456789: U 10.0.34.226:1300 -> 10.0.35.226:5060 INVITE sip:1310123456789@kamailio.local:5060 SIP/2.0. From: <sip:anonymous@10.0.3.44:5060>;tag=26e5ce5850. To: <sip:1310123456789@kamailio.local:5060>. Kamailio forwards request to dispatcher but fails to properly replace the from uri: U 10.0.35.226:5060 -> 10.0.32.36:5060 INVITE sip:+31123456789@kamailio.local:5060 SIP/2.0. From: "anonymous" <sip: +31987654321@10.0.3.44:5060sip:anonymous@anonymous.invalid:5060>;tag=26e5ce5850. To: <sip:0123456789@kamailio.local>. Even though the debug log suggests it did the proper replacement: uac [replace.c:265]: value to store is is '<sip:anonymous@10.0.3.44:5060>' and len is '30' uac [replace.c:268]: Storing in FROM-AVP (for use in reply): '<sip:anonymous@10.0.3.44:5060>' with len '30' uac [replace.c:324]: uri to replace [sip:anonymous@10.0.3.44:5060] uac [replace.c:325]: replacement uri is [sip:anonymous@anonymous.invalid:5060] uac [replace.c:383]: encode is=<AAAAAAAAAAAAAAAAAABQXkFeV15BQUcUXF5AUWxpZDo1MDYw> len=48 uac [replace.c:265]: value to store is is '<sip:1310619024127@kamailio.local:5060>' and len is '54' uac [replace.c:271]: Storing in TO-AVP (for use in reply): '<sip:1310123456789@kamailio.local:5060>' with len '54' uac [replace.c:324]: uri to replace [sip:1310123456789@kamailio.local:5060] uac [replace.c:325]: replacement uri is [sip:0123456789@kamailio.local] uac [replace.c:383]: encode is=<AAAAAAEFAAkGAw0BAANxQlYuERJDCRxeHAoXWgYRXRdbAxtNHxwNCgdACxgubmw6NTA2MA--> len=72 Kamailio version: kamailio 3.3.2 (x86_64/linux) (the latest stable debian/squeeze package). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

3 5

Changing METHOD
by Olle E. Johansson 20 Dec '12

20 Dec '12

Hi! I want to change METHOD on a SIP message from UPDATE to INVITE or REGISTER to INFO. Seems like $rm is read-only. Any other known way to change a requests method? /O

4 8

Send reply with Reason header
by Mino Haluz 19 Dec '12

19 Dec '12

Hi, I know how to use sl_send_reply, but I would like to add "Reason:" header to the reply generated. Is there any command or variable for that? Thank you, Mino

3 3

Video conference with many users
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, I want to do web (HTML5 + WebRTC) Sip client which can do the video conference with multiple users. Current release of SIPML <http://www.sipml5.org/> does 1 to 1 call. I have no idea of conference with many users. Is it the client that we need to modify to accept call and join the conference ? Do I need to send INVITE with extra parameters ? Please advice me. Best Regards, Roy.

3 2

Question registering sip providers in kamailio and forward all?
by Thomas Hoellriegel 19 Dec '12

19 Dec '12

Hi all, I.m. a newbi in Kamailio. My problem: I.m. a blind user and i programmed a voicechat in Freeswitch. 3 have 2 sip providers. vodafone.de sipgate.at and freenet.de. The problem: i must register these proviers in kamailio. And forward all incoming calls to Freeswitch, and outgoing calls from freeswitch to kamailio. Dtmf modes are: auto rfc2833 and inband. Then i call: dorf(a)dorf.blindi.net i like to forward these calls to my freeswitch box. the kamailio ip ist: 217.172.180.108 and the freeswitchbox: 217.172.170.120 Can your help me please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

2 1

NAT Traversal issue
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client->LAN client web page call 2. LAN webrtc client -> 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy.

2 2

New developer: Richard Good
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Richard Good - he is among those developing the IMS extensions for Kamailio (http://code.google.com/p/smile-ims-extensions/), to be merged in the next days to the GIT master branch. His git commit id is: richard.good My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 2

modules_s/timer review
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I was comparing modules_s/timer with modules_k/rtimer to spot the differences. modules_s/timer is working only with core timers (fast and slow timers), while the modules_k/rtimer can start own timer processes, or register a task to main core timer. Other differences ... modules_s/timer: - run on milisecond basis - option to enable/disable timer modules_k/rtimer - run either on second or micro-second basis - option to execute the many route blocks on same timer So, apart of enable/disable timer option, rtimer can do more than what timer module offer (mili-seconds can be run as 1000x micro-second). Enabling/disabling a timer can be workarounded from script, using a shared variable tested at begininning of rtimer route, like: route[RTIMER] { if($shv(rtimer)==0) return; ... } Enabling will mean setting $shv(rtimer)=1, either from other parts of the config or via MI/RPC commands. Now, both modules can be kept, no naming conflict, it is just a matter of code maintenance if we should keep timer module. Enabling/disabling timers can be added to rtimer in the future, but it is unlikely to happen before next major release (unless someone else does it). Is anyone here using modules_s/timer module aware of other differences comparing with rtimer? Looking for comments to keep or obsolete modules_s/timer module ... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.3 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Planning next minor release - v3.3.3
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, I am considering releasing next minor version from branch 3.3 at the beginning of next week, Monday or Tuesday (Dec 17 or 18). For outstanding issues, please use the tracker or send to sr-dev mailing list. Developers that pushed fixed in the master branch should backport them in 3.3 before the next week starts. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

dlg_manage() and add_contact_alias() problem
by Pavel Miskov 18 Dec '12

18 Dec '12

Hi all, I want to end dialog (Kamailio sends BYE to both parties) that is lasting longer then some defined time. everything is working fine when using old fix_nated_contact function, but when using new add_contact_alias function kamailio sends BYE to private addresses. 1. "kamctl fifo dlg_list" output when using fix_nated_contact(): caller_contact:: sip:bob@PUBLIC_IP:50784 callee_contact:: sip:alice@PUBLIC_IP:64472 2. "kamctl fifo dlg_list" output when using add_contact_alias(): caller_contact:: sip:bob@10.2.5.206:10000 callee_contact:: sip:alice@10.2.5.205:20000 dlg_manag() doesn't store contact alias for caller and callee so Kamailio is sending BYE to private addresses. Am I doing something wrong or I can't use possibility of ending dialog with contact aliases? Thank you Pavel

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