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37469 discussions

Re: [Kamailio-Users] Load Balancing Asterisk using Kamailio
by Daniel-Constantin Mierla 27 Apr '09

27 Apr '09

Hello, not cc-ing to mailing list could lead to no answer... private messages from mailing list threads are ignored. Other benefits with mailing lists is that others can answer quicker or better. Thanks, Daniel On 04/27/2009 04:52 PM, Kurt Weasel wrote: > Hi Daniel, thank you for the reply. > I have some more questions to ask. > > 1. Since the dial plan is configured exactly on Asterisk (at > etc/asterisk/sip.conf and /etc/asterisk/extension.conf), do we need to > enable MySQL for Kamailio installation? Or the default installation > configuration will do (without additional dialplan or MySQL setting)? > My idea is only using Kamailio as pure load balancer so we hope to do > this as simple as possible without degrading the function. > > 2. I assume using Kamailio dispatcher module will forward some of the > users to asterik server 1 and some of them to asterisk server 2. Then > if user A is registered with Asterisk server 1 and user B is > registered with Asterisk server 2, can user A call user B? Without > Kamailio, the answer is no since they are in different Asterisk > server. With Kamailio, I am expecting both user A and B can call each > other although they are registered with different servers. Is my > configuration enough to achieve this function? Or do we need some > additional setup for this? > > Thanks again. > > Best Regards, > Kurt > > On Mon, Apr 27, 2009 at 8:02 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 04/27/2009 07:10 AM, Kurt Weasel wrote: > > Hi, > > I searched around the web to load balance asterisk servers and > found Kamailio for possible solution. Let's say I have two > identical asterisk servers with same dialplan and > configuration and I want both servers look like they have same > IP address from clients. Based on my understanding, we need 1 > Kamailio as load balancer and 2 Asterisk servers as the real > servers. Let's say the setup is : > > Kamailio load balancer 192.168.2.1 > Asterisk Server #1 192.168.2.2 > Asterisk Server #2 192.168.2.3 > > My question is, X-Lite softphone Configuration should be set > to domain 192.168.2.1, right? > > I also want to know the step by step configuration to set > kamailio as load balancer. I have not used Kamailio before. > However after searching the documentation, the step (based on > my understanding) is somewhat like this : > > 1. Install Kamailio. I will use the step by step here : > http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn > > 2. Then, using dipatcher module, I will configure it using > this guide here : > http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha > Then modify the dispatcher.list file to match the IP address > of my asterisk servers : > > /1 sip:192.168.2.2:5060 <http://192.168.2.2:5060> > <http://192.168.2.2:5060> > 1 sip:192.168.2.3:5060 <http://192.168.2.3:5060> > <http://192.168.2.3:5060>/ > > > Am I missing some steps? > Do I also need to configure dialplan or any other file at > Kamailio load balancer? Or those two steps basically done it > all for simple load balancing configuration? Thanks for your > responses. > > these steps are ok if you do not deal with NAT. If yes, then > things get a bit more complex, you need PATH support on load > balancer, but AFAIK, Asterisk does not support it -- maybe I am wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] LCR gateway failthrough..
by Nick Warr 27 Apr '09

27 Apr '09

I'm using the LCR module to send traffic based on just the prefix of the number at the moment. I have two gateway groups, one with a local provider to connect locally, one with a larger one to do international calls (they can also connect locally but with a higher price).. When I call next_gw() will LCR use all possible gateway groups, or just the gateway(s) in the first matching group? The second question is tangential to the first, if it does send the traffic to other gateway group after failing through the first, does it regenerate the RURI, because the local provider wants a numerical prefix on the calls I send, and the other one no (basically do I need to account for this in my config?)

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[Serusers] SER with RADIUS
by Leon Li 27 Apr '09

27 Apr '09

Hi, Does anyone still have the instruction of SER RADIUS Howto? The link http://iptel.org/ser/ser_radius.html Found in an old thread doesn't work now. (see below for the old thread) [Serusers] ser RADIUS Howto Jan Janak jan at iptel.org <mailto:serusers%40lists.iptel.org?Subject=%5BSerusers%5D%20ser%20RADIUS %20Howto&In-Reply-To=> Fri Sep 12 02:11:33 UTC 2003 * Previous message: [Serusers] serweb main page display <http://lists.iptel.org/pipermail/serusers/2003-September/002595.html> * Next message: [Serusers] Implementing Voicemail <http://lists.iptel.org/pipermail/serusers/2003-September/002596.html> * Messages sorted by: [ date ] <http://lists.iptel.org/pipermail/serusers/2003-September/date.html#2591 > [ thread ] <http://lists.iptel.org/pipermail/serusers/2003-September/thread.html#25 91> [ subject ] <http://lists.iptel.org/pipermail/serusers/2003-September/subject.html#2 591> [ author ] <http://lists.iptel.org/pipermail/serusers/2003-September/author.html#25 91> ________________________________ Hello, I finished SER RADIUS Howto. The document thoroughly describes installation and configuration of RADIUS server and client libraries for authentication and accounting with ser. RADIUS related questions appear quite often on the mailing lists so I hope the document helps. I would appreciate if anyone using another RADIUS server than FreeRADIUS could describe the setup and send it to me (plaintext is OK, I will convert it to docbook). You can get the document from here: http://iptel.org/ser/ser_radius.html Any feedback is appreciated. Jan. Thanks Leon

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[Kamailio-Users] switch to TCP when UDP message is bigger than MTU
by Sandro Bordacchini 26 Apr '09

26 Apr '09

Hi all. I am doing some test about tcp/udp transport protocols in SIP and IP fragmentation. I would like to test this behaviour, as stated in par. 18.1.1 of rfc3261: If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 [43] congestion controlled transport protocol, such as TCP. If this causes a change in the transport protocol from the one indicated in the top Via, the value in the top Via MUST be changed. This prevents fragmentation of messages over UDP and provides congestion control for larger messages. However, implementations MUST be able to handle messages up to the maximum datagram packet size. To do this, I am running a OpenSer 1.3.2 with a "default" configuration (and very simple: no accounting, no auth, no DB) and I have two SIP phones registered on that server (for example, ext. 100 and ext. 101). When 100 calls 101, the INVITE reaches the OpenSer that appends some "dummy" headers (I have added some append_hf in the route), just to let the message be bigger than the MTU. The INVITE is routed towards 101 with UDP protocol and I have IP fragmentation. Do you have some hint to get the OpenSer work as described in the RFC (i.e. switch automatically to TCP)? Or is this only possible with UACs and not with proxies? TIA -- Ing. Sandro Bordacchini Dexgate Sr. Tech. Engineer P.zza Fermi 1 - Fornacette IT-56012 Calcinaia (Pisa) Tel. +390587424145 ext 580

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[Serusers] Error in CANCEL generation while forking - ser 2.0
by Tomasz Zieleniewski 24 Apr '09

24 Apr '09

Hi, I found an error in the SER proxy behavior. I use SER v_2_0_0 from CVS. When SER forks requests because there is a number of registered users for particular AOR, and then receives first final response, it in the standard way cancels all other transaction. Problems is that it is not compliant with RFC3261 because it doesn't add Max-Forwards header to the CANCEL requests. RFC3261 says that Max-Forwards must be present in any SIP method. CANCEL sip:16@212.180.179.42:9360 SIP/2.0 Via: SIP/2.0/UDP 212.180.179.42:7060;branch=z9hG4bKb721.ceeb7aa5.1 From: <sip:pgr@sip.rd.touk.pl <sip%3Apgr(a)sip.rd.touk.pl>>;tag=374595 Call-ID: 0da793c0ff2e7f2fc81b23e4166bbab3(a)212.180.179.42 To: <sip:tzl@sip.rd.touk.pl <sip%3Atzl(a)sip.rd.touk.pl>> CSeq: 479638 CANCEL Route: <sip:sip.rd.touk.pl:7060;lr> User-Agent: Sip EXpress router(2.0.0 (x86_64/linux)) Content-Length: 0 Kind regards, Tomasz Zieleniewski

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[Kamailio-Users] textops usage
by Juan Perez 24 Apr '09

24 Apr '09

Hello I am receiving the SDP from a provider with this format: g729/8000/1, which I interpret as codec g729, sampling rate 8000 and the number of channels is 1. The calling party [linksys pa2p] apparently does not like the "1" and terminates the call. I want to strip it using textops module and pass back the SDP w/o the "1" and do this in the On_Reply and the Route sections: if (search_body("g729/8000/1")) { xlog("L_INFO","mylog: On Reply 1 section. Found the string.\n"); replace_body("g729/8000/1","g729/8000"); } but it can not even find it. Is leaving me with several questions: 1- is the "re" that I am using right? 2- Do I have to escape the "/"? 3- does the replace and search function go "deep" enough in the SDP? cheers jp

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[Kamailio-Users] Problem with LCR
by Antonio Reale 24 Apr '09

24 Apr '09

Hi all, I'm using kamailio 1.4.4. In kamailio.cfg I have call forward on busy and call forward no answer that works good when I redirect the call to a local user. If I redirect the call to a PSTN number using lcr module I receive the error: ERROR:lcr:next_gw: No ruri_user AVP If I call directly the PSTN number all is OK and I see in the debug: DBG:lcr:next_gw: Added ruri_user_avp <0255XXXXXX> Where I need to set the ruri_user_avp? I just have modparam("lcr", "ruri_user_avp", "$avp(i:500)") in the script. With a very similar configuration and openser 1.2.3 the problem doesn't occur. Thanks. Regards. Antonio.

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[Kamailio-Users] bidirectionnal chating and presence between sip and MSN
by rachid 23 Apr '09

23 Apr '09

Hello all , I want to have a bidirectionnal chating and presence between sip and MSN ,but i dont know how to configure kamailio thank you for your help -- ------------------------------------------------------------ BAHA Rachid Elève ingénieur en 3ème année à l'INPT, Rabat GSM:(+33) 0619787609

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Re: [Serusers] SER Support for SIP Diversion header
by Stefan Sayer 23 Apr '09

23 Apr '09

Hello, o Randy Jackson [04/22/09 19:49]: > Stefan, > > > > For a SER/SEMS configuration using the voicemail app, 4605 calls 4610. > 4610 doesn’t answer and call is forwarded to 9200. If the forwarded > call is routed to SER/SEMS (i.e. see example below) when the SER > (ser-2.0.0-rc1) receives the below INVITE which contains the Diversion > header in the message header portion of the request, what module and > methods do you use to retrieve the parameters from the Diversion header, > like the URI? 9200 does not have a vm account. The vm app will not > answer the call for account 4610 based on the below INVITE. Thanks in > advance. The question is: where do you get the email address from? If it's in DB I think you could get the user from the diversion header into avp with some clever textops and then do the DB query with it to get email address and user into P-App-Param header. If you are only using mode=box (only voicebox, no mail) though, it may be simpler to get the user in SEMS like with this patch to voicemail app: Index: AnswerMachine.cpp =================================================================== --- AnswerMachine.cpp (revision 1354) +++ AnswerMachine.cpp (working copy) @@ -531,10 +531,16 @@ throw AmSession::Exception(500, "voicemail: no email address"); } - user = get_header_keyvalue(iptel_app_param,"usr", "User"); + // user = get_header_keyvalue(iptel_app_param,"usr", "User"); + AmUriParser div_parser; + if (!div_parser.parse_contact(getHeader(req.hdrs, "Diversion"), 0, end)) { + ERROR("could not parse Diversion header"); + throw AmSession::Exception(500, "voicemail: no user found"); + } + user = div_parser.uri_user; if (!user.length()) user = req.user; - + sender = get_header_keyvalue(iptel_app_param, "snd", "Sender"); if (!sender.length()) sender = req.from; btw, isn't Diversion supersedet by History-Info? BR Stefan > > > > > > INVITE sip:9200@192.168.1.1;user=phone > <mailto:9200@192.168.1.1;user=phone> SIP/2.0 > > Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKac816850588;alias > > Max-Forwards: 70 > > From: <sip:4605@192.168.1.2>;tag=1c816838653 > > To: <sip:9200@_192.168.1.1_;user=phone> > > Call-ID: 816837947271200020365(a)192.168.1.2 > > CSeq: 1 INVITE > > Contact: <sip:4605@192.168.1.2;transport=tcp> > > Supported: em,100rel,timer,replaces,path,resource-priority > > Allow: > REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE > > *Diversion: <sip:4610@192.168.1.1>;reason=no-answer;screen=no;privacy=off* > > User-Agent: > > Content-Type: application/sdp > > Content-Length: 270 > > > -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Kamailio-Users] SR-Users-ES maillist
by Iñaki Baz Castillo 23 Apr '09

23 Apr '09

Hi, for the spanish speakers community there is a maillist about Sip-Router: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es This maillist has been merged with the previous Kamailio Users-es maillist, so the maillist history remains. Best regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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