sr-users October 2008

sr-users@lists.kamailio.org

93 participants
169 discussions

[Kamailio-Users] Kamailio/Openser - improve the performance with children process

by Daniel Cardoso Alves

Hi people... Is there a number of children that the Kamailio/Openser for improve the performance? Each process takes care of a request? Thanks in advance. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

15 years, 5 months

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

by luzango mfupe

Hi Ricky,Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango. > > Message: 2 > Date: Tue, 28 Oct 2008 23:10:12 -0700 (PDT) > From: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Subject: [Kamailio-Users] I don't have asterisk audio to openser - > mediaproxy > To: users(a)lists.kamailio.org > Message-ID: <654712.75990.qm(a)web59903.mail.ac4.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, > I have gotten myself an ip it public with my supplier, I have two network > cards in the pc that I am using for openser and mediaproxy together with > asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the > pstn through asterisk I don't have audio, if I call to the pstn they listen > to me well . > > > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7492 ACK > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Length: 0 > > > # > U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060 > INVITE sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Proxy-Authorization: Digest username="112", realm="192.168.10.1", > algorithm=MD5, uri="sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>", > nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", > response="c40a9387fdf5de29115c1edadc7f79db" > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.10.30 > t=0 0 > m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060 > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;branch=z9hG4bKf428b928c25dad03;rport=5060 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > Server: OpenSER (1.3.2-notls (i386/linux)) > Content-Length: 0 > > > # > U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070 > INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 69 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > P-hint: route(3)|setflag7,forcerport,fix_contact > P-hint: inbound->inbound > P-hint: Route[6]: mediaproxy > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.1.64 > t=0 0 > m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received= > 192.168.10.1 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:2685249@192.168.10.1:5070> > Content-Length: 0 > > > I don't have a lot of experience with mediaproxy, and I have some doubts > that such you see they can help me to clarify, inside the file > mediaproxy.ini some options appear which I have configured them but I am not > sure if it is the best way. > > my scenario is the following one: > > <-> UAC<-> NAT <-> ADSL <-> Internet <-> > eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan ( > 192.168.11.1) <-> UAC > > > [MediaProxy] > > start = yes > socket = /var/run/mediaproxy.sock > group = openser > listen = None > allow = None > proxyIP = x.x.x.x (public ip) > ;portRange = 60000:65000 > portRange = 35000:65000 > TOS = 0xb8 > idleTimeout = 60 > holdTimeout = 3600 > forceClose = 0 > > [Accounting] > ; one of none, radius or database > accounting = none > > [Database] > user = dbuser > password = dbpass > host = dbhost > database = radius > table = radacct > > [Radius] > secret = secret > server = localhost > authport = 1812 > acctport = 1813 > dictionaries = /etc/radiusclient-ng/dictionary, > /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary > retries = 2 > timeout = 3 > > > > > this couple of you line inside the openser, I don't still understand them > according to the guide of ser getting started they are for asymmetric > clients, but I don't find an example > > modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") > > somebody that can give me a good help... > > regards > > rickygm > > > > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

15 years, 5 months

[Kamailio-Users] Fwd: Re: How to enable local_route feature ?

by Henning Westerholt

On Wednesday 29 October 2008, Pascal Maugeri wrote: (you can reach the kamailio list at lists.kamailio.org, the lists at openser.org goes to the opensips lists) > Following your instructions I have enable the compilation defined in > Makefile.defs and recompile everything after a make clean. One capture of > the compilation for you to see the define is enable: > > [...] > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott > -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK > [..] > checks.c -o checks.o > [...] > > When I try to start the proxy it complains about: > > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > syntax error > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > invalid local_route statement > > As you announced this feature for OpenSER 1.4 release series, I assume this > is available for kamailio v1.4.2 ? Hi Pascal, i just tried it with the trunk (which should show the same behaviour with regards to this feature). My compilation accepts local_route { [..] } in the configuration. Can you quote the relevant part of your configuration? Cheers, Henning

15 years, 5 months

[Serusers] Error logs for SER

by Shruti Sanadhya

Hi, I am trying to setup a SER on Ubuntu 8.04 (Hardy heron). I have been following the starter's guide for configuring my SER but I do not know where to look for the error logs. I just get a 'bad config file' error. Where can I get more details? Thanks

15 years, 5 months

[Kamailio-Users] error when initilazing "uac.so"

by Alex R.S.M

Hi, I added "uac.so" module to be able to use "uac_replace_from" but got the following error. I tried to initialized the module params I could think of. I am wondering what I am missing . Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:avpops:avpops_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:uac:mod_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:uac:mod_init: 'append_fromtag' RR param is not enabled! - required by AUTO restore mode Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:init_mod: failed to initialize module uac Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:main: error while initializing modules Thanks, Alex _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/

15 years, 6 months

Re: [Kamailio-Users] REGISTER port with Xlite 2.0

by ingdavidcespedes＠cable.net.co

Thanks I'm now trying Zoiper, and it works perfectly, I have solved lot of problems that I thought that were because of my Proxy configuration, but they were because of my X-Lite. Now I will try Twinkle. Thanks for your comments. David Cespedes ----- Mensaje original ----- De: Victor Pascual Ávila <victor.pascual.avila(a)gmail.com> Fecha: Sábado, Octubre 25, 2008 12:51 pm Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > On Sat, Oct 25, 2008 at 12:09 PM, Iñaki Baz Castillo > <ibc(a)aliax.net> wrote: > > El Sábado, 25 de Octubre de 2008, ingdavidcespedes(a)cable.net.co > escribió:> > >> since we don't have an upgrade of X-lite for linux > >> systems, could you recommend me another Softphone. > > > > Always Twinkle softphone, the best one ;) > > I agree-- Twinkle is available for Linux only. > -- > Victor Pascual Ávila > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

15 years, 6 months

[Kamailio-Users] My contrib to the project - Simple UI

by Graham Wooden

Hi list, After seeing that CDRTool was a little too powerful (and complicated for that matter) I decided to take it's CDR gathering and add a "simple" web UI to it allowing me to maintain my SIP users and various settings (some aren't shown in this version right now). Let me preface this - this is very plain, meaning it's not stylized. No CSS, no fancy layouts, etc. Mainly because I have this within my existing UI. However, I stripped it out and is functional as described. Again, for the CDR stuff it does require to have the RADIUS accounting and MySQL setup from CDRTool. Link: https://www.leasedminds.com/ui/ If people see value with this, I will be more than happy to work with someone to make it "stylized" and more featured pack. Enjoy and thanks, -graham

15 years, 6 months

[Kamailio-Users] Proxying media on reinvite

by CSB

We have the following scenario: Nated UAC receives INVITE Call is established, we proxy audio Far end sends Re-INVITE (to inform UAC to use UDPTL) We don't proxy audio so call fails So I need to understand how the appropriate way to force us to proxy the audio when the far end of a reinvite is NATed. Relevant parts of openser.cfg (OpenSER 1.3.2): We're using mediaproxy. modparam("usrloc", "nat_bflag", 6) route { # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, if (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("64.38.1.1:5060;nat=yes"); # insert IP address #xlog("invite and nated so record_route_preset"); #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); xlog("not register and not nated so record_route"); }; #if (method!="REGISTER") { # record_route(); #}; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= #entry in the To header. So if it's loose routed and doesn't have totag #and it's an invite or reply then something's wrong xlog("loose_route"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); xlog("403 in lr"); return; }; if (method=="INVITE") { if (!allow_trusted()) { xlog("!allow_ trusted for r-uri <$ru>"); if (!proxy_authorize("","subscriber")) { proxy_challenge("domain.com","1"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; #check client is nated or that we've already identified it's nated if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); xlog("Recipient is nated so setbflag 6 and use mediaproxy: r-uri <$ru>"); use_media_proxy(); }; }; route(1); return; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message to its destination if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying log(1,"route[1] !t_relay"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); } else { log(1,"route[1] t_relay"); }; } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { xlog("hit onreply_route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); if (isbflagset(6)) { xlog("flag(6) is set for $fu . We're currently in onreply_route(1)"); }; if (isbflagset(7)) { xlog("flag(7) is set for $fu . We're currently in onreply_route(1)"); }; if (status=~"(180)|(183)|2[0-9][0-9]") { xlog("status matches for $fu . We're currently in onreply_route(1)"); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } The re-invite looks like this: INVITE sip:6422334455@58.28.1.1:5061 SIP/2.0 Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 Contact: <sip:022998877@202.180.1.1> Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE User-Agent: UAS Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 349 v=0 o=root 5466 5467 IN IP4 202.180.1.1 s=session c=IN IP4 202.180.1.1 t=0 0 m=image 4724 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy Our OK looks like this: SIP/2.0 200 OK To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport=5060 Record-Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> Contact: 6422334455 <sip:6422334455@58.28.1.1:5061> Server: Linksys/SPA2102-5.1.9 Content-Length: 265 Content-Type: application/sdp v=0 o=- 69685 69685 IN IP4 192.168.1.113 s=- c=IN IP4 192.168.1.113 t=0 0 m=image 16386 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy Unfortunately this code doesn't result in the c record of the OK being updated to the server IP (it is still the private IP of the UAC). In the loose route section, the re-invite doesn't pass (client_nat_test("3") || search("^Route:.*;nat=yes")) so we don't use mediaproxy. In the on-reply route bflag(6) is not set. I don't understand why bflag(6) is not set since the r-uri is to a registered user that has bflag(6) set (at least that's how I understand it). Could anyone point me in the right direction? Thanks Cameron

15 years, 6 months

[Kamailio-Users] [presence] Modifying NOTIFY headers set

by Pascal Maugeri

Hi I need to add a SIP header to outgoing NOTIFY messages, just before it is sent to remote users. Is that possible to do that without using the local_route section (using openser 1.3) ? Regards, Pascal

15 years, 6 months

[Kamailio-Users] [dispatcher] configure the sending of OPTIONS pings

by Pascal Maugeri

Hi How do you configure dispatcher to send OPTIONS ping to load-balanced proxies ? I'm using kamailio 1.4.2 with this of configuration: [...] modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_ping_from", "sip:fend1@10.132.184.33<sip%3Afend1(a)10.132.184.33> ") modparam("dispatcher", "ds_probing_threshhold", 3) [...] The load-balancing works properly : I see the traffic being sent to my proxies. But I don't see any SIP OPTIONS packets. I've read in module doc "This method is only available, if compiled with the probing of failed gateways enabled. " but I can't find how to activate this on compilation time, so I'm not sure if I have this feature or not enable. Regards, Pascal

15 years, 6 months

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sr-users October 2008