sr-users February 2013

sr-users@lists.kamailio.org

84 participants
125 discussions

planning to release Kamailio v3.3.4
by Daniel-Constantin Mierla 25 Feb '13

25 Feb '13

Hello, I am considering packaging a new minor release for branch 3.3 next Monday, on February 25. Please backport any patches that you have to be part of v3.3.4 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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Changing outgoing/source port
by Efelin Novak 25 Feb '13

25 Feb '13

Hi, I'd like to ask whether I am able to change an outgoing port. I'm using t_relay for outgoing INVITEs & Kamailio uses 5060 port defined in listen. When the call fails I am rerouting it using append_branch and I'd like to change an outgoing port to 5061. The reason I'd like to do this is a Loop detection of my buggy VoIP/analog gateway. Am I able to specify an outgoing interface/port for t_relay in Kamailio or should I find an another gateway? :) Thanks for help Kind regards Efelin

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kamailio
by Suhaib Mehyar 25 Feb '13

25 Feb '13

Dears, My name is suhaib mehyar, I tried to install kamailio server, and I successfully set it up, it is locally running fine, both audio and video are very good locally but when I call a client that is connected to the server that is exist after my cisco firewall with private ip and the client is exist after his home router both audio and video are not working. Kindly any advice please??? Best Regards, Suhaib Mehyar Skype : suhaib.mehyar Mob :+962 795 161311

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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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sr-users February 2013