- sr-users - kamailio.org

[Kamailio-Users] controlling inbound calls
by Juan Perez 06 Mar '09

06 Mar '09

hello guys I need to control by IP,port,R-URI who can send calls to our proxy. Looking thru the doc I saw only 1 module that I can use, PERMISSIONS with either Trusted Requests or Address Permissions modes enabled.. Is this the only method or am I missing something? this is for kamailio 1.4.3_no_tls txs a lot jp

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[Serusers] SER and cisco call manager express
by Vittorio Giuffredi 06 Mar '09

06 Mar '09

Hi, I'm relatively new to SER. I'm investigating if a Cisco Router with CME can be connected with a server running SER so I can place a call from my softphone (registered under SER) toward an IP phone that belongs to the Cisco device described above. regards Vittorio Giuffredi ______________________________________________________________________ This email has been scanned by the Candy Group Email Security System. ______________________________________________________________________ Before printing this message, please make sure this is absolutely necessary. Protecting the environment depends on you too! ______________________________________________________________________

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[Kamailio-Users] Normalising
by Brian Chamberlain 06 Mar '09

06 Mar '09

Hi List!, I am trying to write a normalised uri into the radius record for accounting reasons. I have a route[5] that does the transformation for me I translate the uri and store it in a variable. $var(translated_uri) = $ru; $ru = $var(ruri_store); In the accounting module parameters I store Sip-Translated-Request- URI= $var(translated_uri) = $ru; I added the following debug in my route[5] to check the variables are being transformed and they are as I want. xlog("L_INFO", "[TEMP1] Accounting translation: $ru\n"); xlog("L_INFO", "[TEMP2] Accounting translation: $var(translated_uri)\n"); The Sip-Translated-Request-URI is equal to 0 on the radius record.. What am I missing here? I hope I have explained myself well Thanks, Brian

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[Kamailio-Users] Where to add pstn route to .cfg file?
by carl Lougher 06 Mar '09

06 Mar '09

Howdy, I'm getting a little further now but have got stuck at the pstn stage. Now if i delete all the current routes in the kamailio.cfg file and just add a rewritehostport then outbound calls work. Now where exactly do i add this in the .cfg file with all the other routes? I'm not concerned about groups or specfic patterns .* can get sent on. Also for the auth to Trixbox am i better to put as a trusted gw or leave as username/password macth in db? Copy of .cfg file below: [root@zedd kamailio]# cat kamailio.cfg # # $Id: kamailio.cfg 5652 2009-03-02 11:13:38Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=no children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:xxx.xxx.xxx.xxx:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ #modparam("uri_db", "use_uri_table", 0) #modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", #"mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #a#modparam("auth_db", "db_url", #a# "mysql://openser:openserrw@localhost/openser") #a#modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; }; } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ #route[4]{ # force_rport(); # if (nat_uac_test("19")) { # if (method=="REGISTER") { # fix_nated_register(); # } else { # fix_nated_contact(); # } # setflag(5); # } # return; #} # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE) #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

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[Kamailio-Users] t_uac_dlg syntax
by kévin l'heveder 06 Mar '09

06 Mar '09

Hi, Each time I try to use t_uac_dlg from a management script using MI_DATAGRAM (Kamailio v1.4), Kamailio answers "bad headers". I didn't find out some working examples, so i hope someone could help me. Here's what I send on my UDP socket: ":t_uac_dlg:\nCANCEL\nsip:154@192.168.10.250 <nsip%3A154(a)192.168.10.250>\n.\n.\nFrom: sip:655@192.168.10.250 <sip%3A655(a)192.168.10.250>\nTo: sip:154@192.168.10.250 <sip%3A154(a)192.168.10.250>\n" ( \n.\n. : because i don't specify next hop and the socket ; also i don't specify body) If someone could correct me or give me a working example, that would be very nice. Thanks, Koon

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[Serusers] Ser base billing
by Parikhita Behera 06 Mar '09

06 Mar '09

Dera Sir In the SER based Switch , Billing is possible? If possible , what are the option possible like cdr , monthly trafic , termination status etc.... Waiting for your reply Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

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[Kamailio-Users] notls vs tls sources
by Luciano Afranllie 06 Mar '09

06 Mar '09

Hi everybody, May be this is a stupid question but I want to know what is the difference between kamailio-1.x.0-notls_src.tar.gz and kamailio-1.x.0-tls_src.tar.gz I guess it is tls support but INSTALL file of notls version talks about compile with TLS support, that is why I am asking... ... - compile with TLS or SCTP support make TLS=1 make SCTP=1 Thanks Luciano

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[Kamailio-Users] handling of DNS-SRV in kamailio-1.4.3 & 1.5.0
by Klaus Feichtinger 06 Mar '09

06 Mar '09

Hello, I have to relay SIP requests to a special "redundant" destination (= FQDN - e.g. "test.kamailio.loc") that consists of 3 hosts with different priorities. After startup Kamailio creates a DNS-SRV request to the DNS server and receives a response with following answer (e.g.): _sip._udp.test.kamailio.loc: type SRV, class IN, priority 10, weight 100, port 5060, target host1.test.kamailio.loc _sip._udp.test.kamailio.loc: type SRV, class IN, priority 20, weight 90, port 5060, target host2.test.kamailio.loc _sip._udp.test.kamailio.loc: type SRV, class IN, priority 30, weight 80, port 5060, target host3.test.kamailio.loc In worst case - if the prime and secondary host (host1 and host2) are unavailable - I expect that kamailio tries relaying the request to the third host in the list. But it doesn't. It makes retransmission of the original invite to the second host and does not try reaching the third host. For tuning the switchover to the alternative target(2) I have set the module parameter "fr_timer" for TM to 3 seconds (modparam("tm", "fr_timer", 3)). Is this behaviour of Kamailio-1.4.3 and Kamailio-1.5.0 (I've tested it with both versions) as expected (= limited to support only two answers in a DNS SRV reply)? Does anybody have experience with this scenario? Can anybody give me a hint? regards, Klaus F. -- Computer Bild Tarifsieger! GMX FreeDSL - Telefonanschluss + DSL für nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a

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[Kamailio-Users] install kamailio (openser) 1.5.0 from svn
by Daniel-Constantin Mierla 06 Mar '09

06 Mar '09

Hello, for those interested in installing kamailio (openser) 1.5.0 from SVN repository, I have created a new wiki page: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn If you find something missing or some error there, please update the page. Hope it's useful! Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Where are going my pua_dialoginfo publishes ?
by Eric PTAK 06 Mar '09

06 Mar '09

Hi, I'm trying to use *_dialoginfo modules but I'm facing to a missunderstanding, or a bug. In fact, the presence server is working and I'm also able to subscribe to dialog event. But when pua_dialoginfo uses pua to send the Publish, I don't know where it going... I even not sure publishes are really sent. This the log : DBG:dialog:run_create_callbacks: dialog=0xb59d30b0 DBG:pua_dialoginfo:__dialog_created: new INVITE dialog created: from=sip:alice@domain DBG:core:grep_aliases: comparing host [0:domain:0] with us [0:domain:0] DBG:core:grep_aliases: match found for: [0:domain:0] DBG:core:check_self: host == me DBG:pua_dialoginfo:build_dialoginfo: new_body: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:alice@domain"> <dialog id="MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q." call-id="MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q." direction="initiator"> <state>Trying</state> <remote> <identity>sip:bob@domain</identity> <target uri="sip:bob@domain"/> </remote> <local> <identity>sip:alice@domain</identity> <target uri="sip:alice@domain"/> </local> </dialog> </dialog-info> DBG:pua_dialoginfo:dialog_publish: publish uri= sip:alice@domain DBG:pua_dialoginfo:print_publ: publ: DBG:pua_dialoginfo:print_publ: uri= sip:alice@domain DBG:pua_dialoginfo:print_publ: id= DIALOG_PUBLISH.MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q. DBG:pua_dialoginfo:print_publ: expires= 0 DBG:pua:send_publish: pres_uri=sip:alice@domain DBG:pua:search_htable: core_hash= 118 DBG:pua:search_htable: record not found DBG:pua:send_publish: insert type DBG:pua:send_publish: UPDATE_TYPE and no record found DBG:pua:send_publish: request for a publish with expires 0 and no record found DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout DBG:tm:run_reqin_callbacks: trans=0xb59d3270, callback type 1, id 0 entered Regards, Eric.

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[Kamailio-Users] Siremis 0.9.2 with Kamailio 1.5 on Debian 5.0
by mike＠agcusa.com 05 Mar '09

05 Mar '09

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Re: [Kamailio-Users] Newb doco request
by carl Lougher 05 Mar '09

05 Mar '09

Ha ha thanks guys. Who wants the 25cents? Ok i dont have an issue with Asterisk/Trixbox as i've been using that for years. I also dont have an issue with Kamailio installation. The issue i have though is that being a newbie to Kamailio i am struggling to understand and come up with a cfg file to allow the following. 1.To allow Kamailio to accept calls from a specific asterisk/trixbox (authentication would be best). 2.To forward all calls from Kamailio to the Sip provider (how do i set that up? No auth required for sip provider). 3.To apply rtp proxy or nathelper if required. 4.To send cdrs to a mysql database. I've had several attempts at writing stuff to the route part of the cfg file but it always fails to load. Is the best method to find the failure to use /var/log/messages? Thats about it and oh my amazon purchase of "Building Telphony systems with Openser" book turned up today so hopefully reading that might make a difference. Taff.. --- On Thu, 5/3/09, olivier.taylor(a)gmail.com <olivier.taylor(a)gmail.com> wrote: > From: olivier.taylor(a)gmail.com <olivier.taylor(a)gmail.com> > Subject: Re: [Kamailio-Users] Newb doco request > To: "Henning Westerholt" <henning.westerholt(a)1und1.de> > Cc: users(a)lists.kamailio.org > Date: Thursday, 5 March, 2009, 3:36 PM > > > > > > > Pesos, of course :) > > > > Olivier > > > > Henning Westerholt a écrit : > > On Thursday 05 March 2009, olivier taylor wrote: > > > 25 cents bounty, just to understand what's > his problem ;) > > > > euro- or dollar-cent? ;-) > > Henning > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] problem with : arithmetic_operations
by BERGANZ François 05 Mar '09

05 Mar '09

Hello, I need to do arithmetic operations. I could do some test with having looks at : http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#arithmetic_operati ons Simple opérations like $var(a)=1+1 works. But, when we have : $var(n)=0; while ($var(j)<$dbr(nb_max=>rows)){ $var(n) = $var(n) + $dbr(nb_max=>[0,$var(j)]); xlog("L_NOTICE", "log5 $var(n) + $dbr(nb_max=>[0,$var(j)]) \n"); ##à give us log5 0+10 where is the problem? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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Re: [Kamailio-Users] [SR-Dev] git on sourceforge
by Daniel-Constantin Mierla 05 Mar '09

05 Mar '09

IIRC some devels in the past asked for conversion from svn to gitin kamailio (openser), but at that time SF didn't offer it. Andrei started the discussion of how to maintain the parallel source trees for the two projects. We should get to a decision soon. Cheers, Daniel On 03/02/2009 03:01 PM, Jan Janak wrote: > FYI: > > http://apps.sourceforge.net/trac/sitedocs/wiki/Git > > Jan. > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Newb doco request
by carl Lougher 05 Mar '09

05 Mar '09

Hi, I'm after some useful doco to allow me to setup trunks from Trixbox to Kamailio and push all calls to a sip provider. Would possibly need rtpproxy or nathelper as well. Can anyone point me in the right direction as i've been combing the web for days and unable to find anything with good explanations and basic enough for a Kamailio beginner. Thanks, Taff.

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Re: [Kamailio-Users] HOWTO rewrite to: header
by alexander merkulov 04 Mar '09

04 Mar '09

thats work for invite nice, but not work fro cancel even this (very simple form) not change cancel if (method == "BYE" || method == "CANCEL") { # xlog("bye\n");a xlog("L_INFO", "CANCEL - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(t_check_trans()){ xlog("triing to rewrite"); if ( subst('/192/aaa/ig') ) {}; t_relay(); exit; }else{ xlog("no trans"); exit; } 2009/3/4 alexander merkulov <arheops(a)gmail.com> > unfortanly it not work after > location() > known limitation: use non-rewrited message. > > 2009/3/4 Ovidiu Sas <osas(a)voipembedded.com> > > Take a look at textops module: >> http://www.kamailio.net/docs/modules/1.5.x/textops.html#id2467719 >> >> Regards, >> Ovidiu Sas >> >> On Wed, Mar 4, 2009 at 4:20 PM, alexander merkulov <arheops(a)gmail.com> >> wrote: >> > here is correct reqwest and reqwest via kamailio >> > >> > is there any way to rewrite To: field in kamailio >> > because now it nto work >> > >> > correct reqwest(asterisk) >> > RECEIVING FROM: 69.70.173.195:5060 >> > CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 >> > Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a >> > From: "Unknown" <sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com> >> >;tag=as366a0ba0 >> > To: <sip:34249@193.110.78.12:8484> >> > Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com >> > CSeq: 102 CANCEL >> > User-Agent: Asterisk PBX >> > Max-Forwards: 70 >> > Content-Length: 0 >> > >> > incorrect reqwest(multi-homed kamailio, asterisk originate to internal >> eth) >> > >> > RECEIVING FROM: 69.70.173.195:5060 >> > CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 >> > Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0 >> > From: "Unknown" <sip:Unknown@69.70.173.195<sip%3AUnknown(a)69.70.173.195> >> >;tag=as6075df60 >> > Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com >> > To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> >> > CSeq: 102 CANCEL >> > Max-Forwards: 70 >> > asterisk >> > Content-Length: 0 >> > >> > -- >> > Merkulov Alexander >> > >> > _______________________________________________ >> > Kamailio (OpenSER) - Users mailing list >> > Users(a)lists.kamailio.org >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > >> > > > > -- > Merkulov Alexander > -- Merkulov Alexander

3 2

[Kamailio-Users] Redundancy and fault-tolerance
by Uriel Rozenbaum 04 Mar '09

04 Mar '09

Hi gentlemen, I'm trying to find a suitable topology to deploy on our SIP network. We started testing UCARP, but it seems it's not going to do for us. Our topology involves 2 servers with physical IPs on them and 2 floating IPs (to balance load 50% on each). The balancing is manual, we are going to point half our servers (measured in load) to one floating IP and so on. When a server fails, the other will handle the load assuming the other floating IP. We found an issue managing IPs, if we have to make Kamailio listen to a floating IP we have to set: listen=0.0.0.0 and then record_route will just pass that IP on the record route headers, making destination servers not to respond. If we try to use record_route_preset() there, when server A fails, all responses come with the "wrong" IP on record_route when server B assumes A's IP. I'd like to know if some of you found a tidier solution and would like to share some topology and implementation tips. DNS SRV is also on our list, but maybe there is something out there that can help us with this. Thanks in advance, Uriel

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[Kamailio-Users] How to send a to-tag in a CANCEL request generated by carrierroute?
by mayamatakeshi 04 Mar '09

04 Mar '09

Hello, I'm using the module carrierroute. I'm setting the carrierfailureroute table to try a second route if the first one times out. The first route is a GW that sends 183 Ringing with SDP and to-tag. But when the failure_route is called due to timeout and the second route is resolved I see kamailio sending a CANCEL without a to-tag and the GW replies with "481 Call Leg/Transaction Does No Exist". So, This fails to cancel the first request. How can I make kamailio send the to-tag in the CANCEL request? Also, I noticed kamailio sends the INVITE to the second route before sending the CANCEL to the first one. Was this coded on purpose? (I believe this doesn't go against the RFC, but it is something curious). If you bother to look at my cfg, here's the relevant parts: route { .........some stuff ..... DOMAIN_AVP = ""; if (!cr_route("CARRIER_AVP", "DOMAIN_AVP", "$rU", "$rU", "call_id", "HOST_AVP")) { sl_send_reply("404", "No Route"); exit; } t_on_failure("ROUTE_FIRST_ATTEMPT_FAILURE"); if (subst_uri('/(sip:.*);nat=yes/\1/')) { setbflag(NAT_BFLAG); } if (isbflagset(NAT_BFLAG)) { fix_nated_contact(); if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { force_rtp_proxy("cfr"); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("ONREPLY"); if (!t_relay()) { sl_reply_error(); } } failure_route[ROUTE_FIRST_ATTEMPT_FAILURE] { if (isbflagset(NAT_BFLAG)) { unforce_rtp_proxy(); } revert_uri(); if (!cr_next_domain("CARRIER_AVP", "DOMAIN_AVP", "$rU", "HOST_AVP", "$T_reply_code", "DOMAIN_AVP")) { exit; } if (!cr_route("CARRIER_AVP", "DOMAIN_AVP", "$rU", "$rU", "call_id", "HOST_AVP")) { exit; } t_on_failure("ROUTE_SECOND_ATTEMPT_FAILURE"); append_branch(); route(RELAY_REQUEST); } failure_route[ROUTE_SECOND_ATTEMPT_FAILURE] { if (isbflagset(NAT_BFLAG) ) { unforce_rtp_proxy(); } }

2 2

[Kamailio-Users] htable question
by Klaus Darilion 04 Mar '09

04 Mar '09

Hi! Sorry for not knowing how hash table works, thus my questions may be a bit stupid: How many items can be stored in hashtable? Is it limited to "size" parameter, e.g. 10 means max. 1024 entries? Or is it as much as memory is available (what for is the size parameter in this case)? How can I delete a key from htable? Example: I want to track concurrent calls (lets pretend there is no dialog module :-): # pseudo language if INVITE $sht(a=>$ci) = $ts; ... elseif BYE $avp(s:duration) = $ts - $avp(s:duration); # how to delete this key now from the htable? Is it possible to iterate over all entries in the htable? thanks klaus

2 1

Re: [Serusers] Need easiest way to set/clear a flag and refer to it later in same call session
by Miklos Tirpak 04 Mar '09

04 Mar '09

Please read the getting started documentation from http://www.iptel.org/doc. This is a bit old, and I strongly suggest to use ser 2.x instead of 0.9.x, therefore some additional documents might be also useful: http://www.iptel.org/ser/doc/010whatsnew. Regards, Miklos p.s. Please post all the questions/answers to the serusers list, others might be interested as well. On 03/03/2009 08:35 PM, Martin Lane wrote: > Where can I learn the syntax or regexp format of the ser config file? > > On Thu, Feb 26, 2009 at 3:50 AM, Miklos Tirpak <miklos(a)iptel.org> wrote: > >> Hello, >> >> for simple and very fast boolean checks you can use the flags: >> >> - declare the flag(s) at the beginning of the config: >> >> flags >> FLAG_FOO, >> FLAG_BAR; >> >> - Call the setflag(FLAG_FOO) and resetflag(FLAG_FOO) functions to set and >> clear the flag when the INVITE is processed. >> - Use "if (isflagset(FLAG_FOO)) {...}" to check the state of the flag in >> the reply route. >> >> For storing integer or string values you can use AVPs: >> >> # during INVITE processing >> $foo = "myvalue"; >> $bar = 2; >> >> # in reply_route: >> if ($foo == "myvalue") {...} >> if ($bar) {...} >> >> Have a look at etc/ser-oob.cfg for examples. The values are attached to the >> transaction in both cases, therefore you need to forward the request with tm >> module. >> >> Miklos >> >> >> On 02/24/2009 06:40 AM, Frank Durda IV wrote: >> >>> I need a simple way to set either a boolean flag (or integer >>> value) during the INVITE of a given call in ser.cfg, and test >>> that value via "if" elsehere in ser.cfg during the 183/200 >>> response messages for that same call. >>> This would be a data object unique to a given call, >>> which should disappear/lose persistence/end scope when >>> a given call ends. Not global, no need to write the >>> value to disk, doesn't need to survive a restart of SER, etc. >>> >>> The various SER documents describe a number of things that >>> could do this, but most seem overly complex, storing the object >>> in a MYSQL database on disk or something, and these all seem >>> so excessive. There also appear to be a few built-in flags, >>> but the documentation suggests that all are actually there >>> for specific uses and so use for other things might not work. >>> >>> I simply want to make a determination of something during >>> the INVITE for call 12345, and be able to refer to that >>> flag or integer value later on when the various response >>> messages for the same call (#12345) are coming back >>> through. I can't make the same determination on each >>> message after the INVITE because the original src_ip >>> address value where the INVITE came from doesn't seem >>> to be in a testable variable when handling reply >>> messages, or at least not in one that is clearly >>> documented. >>> >>> >>> I've chased a number of possible ways this might be done, >>> but tire of being referred away from the doc that came with >>> the source and over to web sites that tell me this or that module >>> that might do part of what I need is deprecated and to use >>> this other module, but the web site for the other module says >>> that the first module is better suited to what I want to >>> do (a circle), doesn't show how the set value can actually >>> be tested, or the web page just doesn't work anymore. >>> >>> So, assuming there is a simple way to do this, I need a >>> quick code snip showing how to set/clear/assign a value to such >>> a beast within INVITE handling, and a quick code snip of how >>> to test it in an "if" statement during 183/200/110 type >>> response messages. Whatever it is, the value set, cleared >>> or being tested in one call should not affect any other >>> call. >>> >>> >>> Thanks in advance! >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >

1 0

[Kamailio-Users] how to suppr a SDP param
by BERGANZ François 04 Mar '09

04 Mar '09

Hello, In my SDP, I have Media Attribute (a): silenceSupp:off - - - - How can I suppr it ?because, I have a phone wich have error when it have that! Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 1

[Kamailio-Users] Trixbox>openser is this possible?
by carl Lougher 04 Mar '09

04 Mar '09

Howdy, Is it possible to connect a trixbox trunk to openser and then flick the cdr to another box and allow rtp streams to go direct from customer trixbox to sip provider? What cdr billing methods are best from openser? I would like to dump in a mysql database if poss? Thanks, Taff..

2 1

[Kamailio-Users] Kamailio (OpenSER) v1.5.0 Released
by Daniel-Constantin Mierla 03 Mar '09

03 Mar '09

Hello, Kamailio (OpenSER) 1.5.0 has been released. It represent the major version with the biggest new features boost since project was started back in 2005. Release 1.5.0 comes after a shorter development period than usual - about five months of coding and more than one month of extensive testing, but brings lot of new features and improvements, with special care to stability, performance, security and flexibility. A short summary of what the release brings out: * configuration file caching system - generic storage engine in memory via htable module allowing to share data between processes and reduce SQL operations by storing results in cache * execute SQL queries against multiple databases and access the results in a fast and natural fashion * execute config file route blocks on timer * send SIP requests from config file * perform HTTP queries from config file - integration with web service * dialog info presence support - RFC4235 * support for TLS server_name extension (aka SNI=server name indication) * support for non-blocking logging * many enhancements to least cost routing (LCR), load balancing (dispatcher) and carrier routing (carrierroute) * access in config file to both request or reply during transaction processing * perl-like regular expression support (PCRE library) * sanity check of SIP message structure * protection against SIP reply attack * twelve new modules * code restructuring in core - slimmer and less exposed to issues * web management interface - SIREMIS - http://siremis.asipto.com You can download the tarball of the released sources at * http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions you can find at * http://www.kamailio.org/pub/kamailio/latest/bin/ * http://www.kamailio.org/pub/kamailio/latest/packages/ Packages will be uploaded as soon as they are submitted. Documentation: * modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ * migration guide: http://www.kamailio.org/dokuwiki/doku.php/install:1.4.x-to-1.5.0 Full release notes at: * http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes Many thanks to all that made this release possible: the management board, developers, testers and community. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Carrierrout issue
by Ali Jawad 03 Mar '09

03 Mar '09

Hi I do use carrieroute to balance calls to 3 servers and those 3 servers do use a2billing and they forward calls to my pstn gw. I do have a graph that has the active calls on them randomly and it does not depend on the load at hand the calls spike up on all 3 servers together. And most of those calls are in ringing state but with no callerID. Normally in the log for carrieroute I have something like this: Mar 3 13:39:10 [4288] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxx ACC: transaction answered: timestamp=1236087551;method=INVITE;from_tag=rMciw2o;to_tag=as7761f5cf;ca ll_id=c745983c;code=200;reason=OK ACC: request acknowledged: timestamp=1236087551;method=ACK;from_tag=rMciw2o;to_tag=as7761f5cf;call_ id=c745983c;code=200;reason=OK Mar 3 13:39:11 [4286] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxxx Mar 3 13:39:12 [4287] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxx ACC: transaction answered: timestamp=1236087552;method=INVITE;from_tag=yJ+hSWA;to_tag=as26649b96;ca ll_id=a1c42904;code=183;reason=Session Progress ACC: transaction answered: timestamp=1236087552;method=INVITE;from_tag=CAJpAXg;to_tag=as5d24ff4f;ca ll_id=a35563d5;code=183;reason=Session Progress ACC: transaction answered: timestamp=1236087552;method=INVITE;from_tag=7i2kutG;to_tag=as6c125d0e;ca ll_id=2739b561;code=183;reason=Session Progress Mar 3 13:39:12 [4285] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxxxxxx ACC: transaction answered: timestamp=1236087553;method=INVITE;from_tag=tMs9y8M;to_tag=as18b7f087;ca ll_id=7d65d34b;code=183;reason=Session Progress However when this problem happens the log screen only has the something like the below and as said the calls are in ringing state with no caller ID: Mar 3 13:39:11 [4286] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxxx Mar 3 13:39:12 [4287] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxx

2 3

[Kamailio-Users] preparing 1.5.0 release
by Daniel-Constantin Mierla 03 Mar '09

03 Mar '09

Hello, preparations for 1.5.0 release starts. Any developer that has to commit any code, please do it quickly. Branch will be created very soon. Many thanks for those helping with development and testing. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] 1.5.0 is awesome.
by Alex Balashov 03 Mar '09

03 Mar '09

Thanks, everyone! Very useful and excellent new features. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

2 1

[Kamailio-Users] sqlops problem
by BERGANZ François 03 Mar '09

03 Mar '09

I follow the module instructions and I do : sql_query("ca", "select server from servers where etat_ping=oui & etat_asterisk=oui", "nb_srv"); $var(i) = 0; $var(s)=$dbr(nb_srv=>[server,$var(i)]); ### line 166 I have that error: ERROR:sqlops:sql_parse_index: bad number <server> ERROR:sqlops:pv_parse_dbr_name: invalid index in [[server,$var(i)]] ERROR:core:pv_parse_spec: pvar "dbr" has an invalid name param [nb_srv=>[server,$var(i)]] ERROR:core:pv_parse_spec: wrong char [)/41] in [$dbr(nb_srv=>[server,$var(i)])] at [29 (5)] CRITICAL:core:yyerror: parse error in config file, line 166, column 110-140: unknown script variable Anyone know why? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 1

[Kamailio-Users] How to change request uri with Header value?
by Ramu 02 Mar '09

02 Mar '09

Hi All, I have situation where I need to change the request uri with To header value. Scenario: OpenServ will receive INVITE as follows INVITE sip:2000@open-ser-ip SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. Now I would like to change the request uri as follows: INVITE sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com> SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. and will do a lookup location. Ramu

3 5

Re: [Kamailio-Users] Handling CANCEL and 487
by Daniel-Constantin Mierla 02 Mar '09

02 Mar '09

Hello, ok. I am cc-ing to mailing list to close the discussion thread. The issue was actually caused by a mis-configuration. Cheers, Daniel On 03/02/2009 05:53 PM, Juan Perez wrote: > Thank you Daniel, I applied that code and it is working now. > I appreciate it a lot. > cheers > jp > > ------------------------------------------------------------------------ > *From:* Juan Perez <jperezsip2008(a)yahoo.com> > *To:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *Sent:* Friday, February 27, 2009 1:27:55 PM > *Subject:* Re: [Kamailio-Users] Handling CANCEL and 487 > > thank you a lot, I will do it and let you know > question: if the call is not cancelled, after the 200 OK there is an > ACK, why that ACK does not have any prob? > > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *To:* Juan Perez <jperezsip2008(a)yahoo.com> > *Sent:* Friday, February 27, 2009 11:36:17 AM > *Subject:* Re: [Kamailio-Users] Handling CANCEL and 487 > > > > On 02/27/2009 06:32 PM, Juan Perez wrote: > > thank you a lot, I am attaching the config file > in the main route block you have nothing that process the ack, so it > gets drop. > > Try to add: > > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > t_relay(); > exit; > } else { > exit; > } > } > > Daniel > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] fix_nated_contact onreply
by samuel 02 Mar '09

02 Mar '09

I have seen lots of default config files where in the reply route only after checking the message (client_nat_test(1)) fix_nated is called. Why is not called when the NAT flag is set upong lookup_XX? In ser-oob I think the reply_route should include the case of a user called behind a NAT and the reply is not fixed due to some router in the middle. Will it hurt including fix_nated_contact in the case of checking the flag? Friday thoughts... Samuel

2 4

Re: [Kamailio-Users] how to know if my gw servers are active ?
by BERGANZ François 02 Mar '09

02 Mar '09

For the database, asterisk and SER work on the sameone. in sip.conf create one for SER and insecure=invite. It accept all from SER In the SER, configure the auth_db module to check the username col and password col. SER is a wall between client and asterisk(s). The same SER can be a wall between asterisk and provider. It is just a gymnastic! To test, it, the best is to have all in the same UC. And have one virtual networkInterface per Asterisk and SER. Cordialement, BERGANZ François http://www.acropolistelecom.net  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Abdul Hakeem [mailto:alhakeem100@hotmail.com] Envoyé : lundi 2 mars 2009 12:14 À : 'BERGANZ François' Objet : RE: [Kamailio-Users] how to know if my gw servers are active ? Hi, Thanks for the prompt reply. My comments are below: -----Original Message----- From: BERGANZ François [mailto:francois@acropolistelecom.net] Sent: 02 March 2009 07:59 To: alhakeem(a)ipextelecom.net Cc: users(a)lists.kamailio.org Subject: RE: [Kamailio-Users] how to know if my gw servers are active ? Abdul Hakeem, I use Asterisk-SER for another way. SER authenticate users and asterisk(s) accept all SIP frames from the SER. >>How did you create the users databases on Asterisk and Ser ? Also, how did you configure Asterisk to forward registration requests to Ser ? If you could send me sample Asterisk/Ser configuration, I'll be much grateful. Then, asterisk(s) forward the call to the provider. When a call come from the provider, asterisk forward it to the SER and the SER just do a lookup(location)! >>What happens if a call comes in to SER directly, & it has to do a quick lookup to see which Asterisk the user is currently located ? The reason I ask is because for example User-A with e164 number/DID +447931800952 might be registered to Asterisk-X now and 10 minutes later register with Asterisk-Y, Did you implement some refresh mechanism on the SER database ?. Because in this case inbound IP calls are hitting SER first before being redirected to the appropriate Asterisk server. I look forward to your reply. Regards, Abdul Hakeem Cordialement, BERGANZ François ? Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Abdul Hakeem [mailto:alhakeem@gmail.com] Envoyé : samedi 28 février 2009 09:41 À : 'BERGANZ François' Cc : users(a)lists.kamailio.org Objet : RE: [Kamailio-Users] how to know if my gw servers are active ? Hello, How did you setup your Asterisk to update SER/OpenSER/Kamalito's database ? I am trying to setup a solution for 2 Asterisk or more servers to allow dynamic and not static user registration. i.e. any user can register with any of the Asterisk servers. Let's say if UserA register with Asterisk1, this will reflect on SER's database where to route the call. In this scenario of UserA now register with Asterisk4 5 minutes later, SER's database will be automatically updated and calls to UserA will be automatically routed to Asterisk4. What I am hoping to achieve is not to do this via manual intervention. Any this will be much appreciated. Cheers, Abdul Hakeem -----Original Message----- From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of BERGANZ François Sent: 27 February 2009 10:24 To: 'Daniel-Constantin Mierla' Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] how to know if my gw servers are active ? Yes, my asterisk update a database to help the SER Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 27 février 2009 11:19 À : BERGANZ François Cc : 'Henning Westerholt'; users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] how to know if my gw servers are active ? On 02/27/2009 12:10 PM, BERGANZ François wrote: > I need to dispatch to the asterisk which have less users connected > How do you get the asterisk with the less users connected? You have some data in a db table? Cheers, Daniel > > Cordialement, > BERGANZ François > P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Henning Westerholt [mailto:henning.westerholt@1und1.de] > Envoyé : vendredi 27 février 2009 11:07 À : BERGANZ François Cc : > users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] how to know if > my gw servers are active ? > > On Friday 27 February 2009, BERGANZ François wrote: > >> If I cant use dispatcher module? >> Because, I don’t use the module algo to choose the dst. >> > > Well, then perhaps you need to use another solution, or adapt the dispatcher > code to your needs. May i ask what is the problem with existing algorithms? > What exactly do you want to achive? > > Cheers, > > Henning > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] how to know if my gw servers are active ?
by BERGANZ François 02 Mar '09

02 Mar '09

hello, I have a list of servers in my database. I need that openser send OPTIONS to servers. for example, I could have a col 'active' to know if each server is active or not. In fact, I need to do an other style of dispatch, and, I cant use the dispatcher module! is there a solution for that ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

4 14

[Kamailio-Users] CANCEL over nat
by alexander merkulov 01 Mar '09

01 Mar '09

i have async rtp proxy setup on multihomed host. config for transfer cancel and invites: if(lookup("location")){ log("found loc"); fix_nated_contact(); $avp(i:11)=$fU; $var(fr)="sip:"+$avp(i:11)+"@69.70.173.195"; xlog("replace $var(fr)"); if (method == "BYE" || method == "CANCEL"){ unforce_rtp_proxy(); # force_rtp_proxy("AIEOC"); }else{ uac_replace_from("$avp(i:11)","$var(fr)"); if(force_rtp_proxy("FAIEOC")) t_on_reply("2"); } route(1); exit(); } cancel is routed ok, ATA get cancel in form of (ip changed): RECEIVING FROM: 69.70.xxx.xx:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as704e6b0d Call-ID: 735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180 To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 but ignore it. same with any other ATA type(including softphones) can u please explaine me what is incorrect in CANCEL? cancel produced by asterisk.CALL-id is correct. here is corespondoing invite 16:06:56.0 RECEIVING FROM: 69.70.173.195:5060 INVITE sip:34249@193.110.78.12:8484 SIP/2.0 Record-Route: <sip:69.70.173.195;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--> Record-Route: <sip:192.168.2.170;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--> Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0 Via: SIP/2.0/UDP 192.168.1.180:5060;rport=5060;branch=z9hG4bK49fa4d6d From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as704e6b0d To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> Contact: <sip:Unknown@192.168.1.180:5060> Call-ID: 735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Sun, 01 Mar 2009 14:05:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 447 v=0 o=root 8300 8300 IN IP4 69.70.173.195 s=session c=IN IP4 69.70.173.195 t=0 0 m=audio 10336 RTP/AVP 0 8 110 97 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes -- Merkulov Alexander

1 0

[Kamailio-Users] dial the prefix #31# to mask the number
by ALAEDDINE abbech 01 Mar '09

01 Mar '09

Hi When i dial #31# 011111with a sip phone i have an INVITE in OPENSER with a number %2331%23 011111. Who know how can i send an INVITE whith prefix #31# using a phone or softphone. thanks,

14 47

Re: [Kamailio-Users] [OpenSIPS-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
by Antonio Reale 27 Feb '09

27 Feb '09

Sorry for the delay. I did not read your reply. Nice solution. I need to check that IOS based pbx can read a custom header. Regards. Antonio. > -----Original Message----- > From: users-bounces(a)lists.opensips.org [mailto:users- > bounces(a)lists.opensips.org] On Behalf Of Iñaki Baz Castillo > Sent: Saturday, February 21, 2009 3:42 AM > To: users(a)lists.opensips.org > Subject: Re: [OpenSIPS-Users] [OpenSER-Users] [OT] How to handle different > DID's in incoming calls for a registered client? > > El Viernes, 20 de Febrero de 2009, Antonio Reale escribió: > > Hi Iñaki, > > did you find a way to solve this problem. I have the same problem with > > devices that can't read the To: value. > > For customers with 10/100 assigned number I routed the entire root number > > to the customer's IP (loosing auth...), but for users with only one > > supplementary number it is not practicable. > > Has someone a suggestion for this problem? > > Thanks to all. > > Regards. > > In the proxy I add a custom header: > P-Called-Number: +34123123123 > so the pbx receiving this INVITE can read that header. > > -- > Iñaki Baz Castillo > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users

4 3

[OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
by Iñaki Baz Castillo 27 Feb '09

27 Feb '09

Hi, maybe this question is a bit off-topic so I'm sorry for that. My question is about SIP providers using OpenSer that associate PSTN numbers to their local clients (SIP accounts): Usually the client must register to OpenSer in order to receive calls. Then it will appear in location table with "Contact=sip:clientX@IP". Suppose clientX has two PSTN numbers associated in a ENUM entry: +34999000111 +34999000222 When anyone in PSTN world calls to +34999000222 the call will arrive to the OpenSer from a gateway in an INVITE like: INVITE sip:+34999000222@gateway SIP/2.0 To: <sip:+34999000222@gateway> The OpenSer will do the lookup in location table and finally send this INVITE to the clientX: INVITE sip:clientX@IP_clientX SIP/2.0 To: <sip:+34999000222@gateway> The info about the called PSTN number is just available in "To" header, so a way to get different behaviour for each associated PSTN number is matching "To" URI. Is common to do it? which other alternatives are there? Thanks for any comment. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

7 17

Re: [Kamailio-Users] ACC with mysql
by Daniel-Constantin Mierla 27 Feb '09

27 Feb '09

According to the config, you do not even set the accounting flag for ACK, or I missed something there? Can you compile with memory debug turned on and test? Watch the syslog for critical errors. It can turn to be a buffer overflow somewhere. Once again, what is the version are you using? Cheers, Daniel On 02/27/2009 01:45 PM, Uriel Rozenbaum wrote: > Here it is > > On Fri, Feb 27, 2009 at 9:31 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 02/27/2009 12:57 PM, Uriel Rozenbaum wrote: > > I´ve checked the captures again and all ACK messages have the > RPID, for 487 or 200 messages. > > ok, but is the ack you sent causing the trouble? It does not look > as an ack for 200ok, because it does not have a Route header. > Non-200ok ACK do not get accounted. > > > I'm just using a plain build of kamailio downloaded from the > site including carrerroute module. > > Would it be useful if I sent the .cfg? > > > Could help, yes. > > Daniel > > > On Fri, Feb 27, 2009 at 8:34 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com> > <mailto:miconda@gmail.com <mailto:miconda@gmail.com>>> wrote: > > I tried to reproduce, but no success. The crash is > indicated when > processing the lumps, a very old piece of code, so I guess > there > is somewhere a buffer overlow. > > Are you having some private modules that are you using the > config > file? > > Thanks, > Daniel > > > > On 02/27/2009 11:39 AM, Daniel-Constantin Mierla wrote: > > Hello, > > is this of an ACK due to 200ok? Are you doing > record-routing? > Because Route header is missing. > > If you don't do record-routing, then the ACK should be > end-to-end between caller and callee and you cannot > account. > > Anyway, the reported issue should be fixed. > > Cheers, > Daniel > > > On 02/24/2009 10:05 PM, Uriel Rozenbaum wrote: > > Daniel, I've been trying a litte longer with the server > and I captured one ACK that definetely has RPID: > > /ACK sip:1054111556446967@cc1int.x.com.ar > <mailto:sip%3A1054111556446967@cc1int.x.com.ar> > <mailto:sip%3A1054111556446967@cc1int.x.com.ar > <mailto:sip%253A1054111556446967@cc1int.x.com.ar>> > <mailto:sip%3A1054111556446967@cc1int.x.com.ar > <mailto:sip%253A1054111556446967@cc1int.x.com.ar> > <mailto:sip%253A1054111556446967@cc1int.x.com.ar > <mailto:sip%25253A1054111556446967@cc1int.x.com.ar>>> SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.200.11:5060;branch=z9hG4bK31a69f48;rport > From: "541160911100" > <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>>;tag=as799b3334 > > To: <sip:1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar> > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar>> > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar> > > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%25253A1054111556446967@cc1int.cpsnet.com.ar>>>>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.90f4 > > > > Contact: <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>> > > Call-ID: > 5d7adcce1399bafd45c194b049479a1b(a)192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11>> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11>>> > CSeq: 102 ACK > User-Agent: Metrotel > Max-Forwards: 70 > Remote-Party-ID: "541160911100" > <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>>;privacy=off;screen=no > > > Content-Length: 0/ > > Hope it helps, > Uriel > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] ACC with mysql
by Uriel Rozenbaum 27 Feb '09

27 Feb '09

Hey guys, I'm using kamailio-1.4.3-notls with ACC module on mysql. I'm trying to benchmark it installed on a virtual server (Xen on CentOS) and I have everything going OK for my basic configuration but when I turn on ACC (by flagging transactions) I get the following lines on the log: Feb 19 08:48:57 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: child process 4899 exited by a signal 11 Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: core was generated Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: terminating due to SIGCHLD Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4902]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4903]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4901]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4897]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4904]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4898]: INFO:core:sig_usr: signal 15 received This issue only happens when i account messages using the following lines: if (is_method("INVITE|BYE|CANCEL")) { setflag(1); setflag(2); } and the configuration for the modules are ass follows: # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # Descomentar para guardar parametros adicionales # modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_extra", "cseq=$cs;privacy=$dip;reason=$dir;from_uri=$fu;from_username=$fU;from_displayname=$fn;from_domain=$fd;req_username=$rU;req_domain=$rd;rpid=$re;source_ip_addr=$si;source_port=$sp;to_uri=$tu;to_username=$tU;to_displayname=$tn;to_domain=$td;original_uri=$ou;original_uri_username=$oU;original_uri_domain=$od;unix_timestamp=$Ts") Do you think there is something I'm missing on the configuration? I tried to modify parameters on mysql but get the same errors all the time? Thanks in advance! Uriel

3 17

Re: [Kamailio-Users] how to know if the next gw is up?
by Daniel-Constantin Mierla 27 Feb '09

27 Feb '09

Please direct your questions to users mailing lists. Private messages of such type are not honored very fast, they usually end in my very long unread mails pool. See the dispatcher module, maybe it helps you. Thanks, Daniel On 02/27/2009 11:18 AM, BERGANZ François wrote: > Hello, > > I have 1SER and 5 Asterisk. > I need to choose the dst with my algo... > But, I need to have my server list and choose a server which is online. > The best thing could be to have OPTIONS send to my asterisk and write responses in the database! > > Have you an idea? > > > Cordialement, > BERGANZ François >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > -----Message d'origine----- > De : users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] De la part de Daniel-Constantin Mierla > Envoyé : vendredi 27 février 2009 09:10 > À : c_lougher(a)yahoo.co.uk > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] Kamailio Newb questions > > > > On 02/27/2009 12:07 AM, carl Lougher wrote: > >> So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy? >> >> > nathelper/rtpproxy does rtp relay on server, so the rtp goes: [caller] > === [rtpproxy] === [callee] > > In case of dealing with symmetric NAT, it is the only feasible way now > to make it work. > > If STUN is used, the rtp goes: [caller] === [callee] . But does not work > for symmetric nat. > > The best is to combine both of them so you get as less as possible RTP > relaying on server. > > Cheers, > Daniel > > >> What are you using rtpproxy for that is different than stun? >> >> >> --- On Thu, 26/2/09, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> >> >> >>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>> Subject: Re: [Kamailio-Users] Kamailio Newb questions >>> To: c_lougher(a)yahoo.co.uk >>> Cc: users(a)lists.kamailio.org >>> Date: Thursday, 26 February, 2009, 12:27 PM >>> On 02/26/2009 01:19 PM, carl Lougher wrote: >>> >>> >>>> Thanks for that. So does it mean by using rtpproxy you >>>> >>>> >>> will therefore carry all the rtp streams through that server >>> yes, that is the role of RTPProxy - to proxy the RTP >>> streams, therefore those go via the server. >>> >>> If you want end-to-end RTP stream, then look at STUN, if >>> the phones are not behind symmetric nat, it can help. >>> >>> >>> >>>> or can it be redirected to the sip provider from the >>>> >>>> >>> endpoint? >>> >>> >>>> Also how do you put the kamailio server in the >>>> >>>> >>> equation? Do you set it up as an external proxy for the >>> clients or do you register the clients to it then just use >>> asterisk for the media/vmail etc? >>> >>> >>>> >>>> >>>> >>> I do everything in kamailio but the media services which i >>> do with asterisk (vmail, ivr, ...) - authentication, >>> registration, call routing is done in kamailio. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>>> --- On Thu, 26/2/09, Daniel-Constantin Mierla >>>> >>>> >>> <miconda(a)gmail.com> wrote: >>> >>> >>>> >>>> >>>> >>>>> From: Daniel-Constantin Mierla >>>>> >>>>> >>> <miconda(a)gmail.com> >>> >>> >>>>> Subject: Re: [Kamailio-Users] Kamailio Newb >>>>> >>>>> >>> questions >>> >>> >>>>> To: c_lougher(a)yahoo.co.uk >>>>> Cc: users(a)lists.kamailio.org >>>>> Date: Thursday, 26 February, 2009, 9:16 AM >>>>> Hello, >>>>> >>>>> On 02/26/2009 12:59 AM, carl Lougher wrote: >>>>> >>>>> >>>>> >>>>>> Howdy, >>>>>> I'm trying to remove the media/rtp streams >>>>>> >>>>>> >>> from an >>> >>> >>>>>> >>>>>> >>>>>> >>>>> asterisk server for natted users so would like to >>>>> >>>>> >>> know if >>> >>> >>>>> this is possible with kamailio. >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> yes it is possible. nathelper+rtpproxy is the >>>>> >>>>> >>> option I use >>> >>> >>>>> and prefer because of flexibility and >>>>> >>>>> >>> performances. You can see an >>> >>> >>>>> example at: >>>>> >>>>> >>>>> >>> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy >>> >>> >>>>> >>>>> >>>>> >>>>>> Qu's: >>>>>> What is the best option? >>>>>> rtpproxy/mediaproxy? >>>>>> nathelper? >>>>>> >>>>>> If i use kamailio to achieve this does it mean >>>>>> >>>>>> >>> that i >>> >>> >>>>>> >>>>>> >>>>>> >>>>> still have to carry the rtp streams through the >>>>> >>>>> >>> kamailio >>> >>> >>>>> server instead? >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> through the rtpproxy server, which can be located >>>>> >>>>> >>> on same >>> >>> >>>>> or different machine than kamailio. >>>>> >>>>> >>>>> >>>>> >>>>>> Also will i need to change the logon info for >>>>>> >>>>>> >>> the >>> >>> >>>>>> >>>>>> >>>>>> >>>>> clients so they now logon to kamailio then i just >>>>> >>>>> >>> point >>> >>> >>>>> registrar to asterisk? >>>>> >>>>> >>>>> >>>>>> Can i use kamailio for sip trunks to asterisk >>>>>> >>>>>> >>> and >>> >>> >>>>>> >>>>>> >>>>>> >>>>> carry rtp and natted clients media streams rather >>>>> >>>>> >>> than >>> >>> >>>>> register to asterisk? >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> Yes, you can register to kamailio, see registrar >>>>> >>>>> >>> and usrloc >>> >>> >>>>> modules. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> >>>>> >>>>> >>>>>> Many thanks, >>>>>> Taff.. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>> _______________________________________________ >>> >>> >>>>>> Kamailio (OpenSER) - Users mailing list >>>>>> Users(a)lists.kamailio.org >>>>>> >>>>>> >>>>>> >>>>>> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >>>>> >>>>> >>>>> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> -- Daniel-Constantin Mierla >>>>> http://www.asipto.com >>>>> >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (OpenSER) - Users mailing list >>>> Users(a)lists.kamailio.org >>>> >>>> >>>> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >>>> >>>> >>>> >>> -- Daniel-Constantin Mierla >>> http://www.asipto.com >>> >>> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Kamailio Newb questions
by carl Lougher 27 Feb '09

27 Feb '09

Howdy, I'm trying to remove the media/rtp streams from an asterisk server for natted users so would like to know if this is possible with kamailio. Qu's: What is the best option? rtpproxy/mediaproxy? nathelper? If i use kamailio to achieve this does it mean that i still have to carry the rtp streams through the kamailio server instead? Also will i need to change the logon info for the clients so they now logon to kamailio then i just point registrar to asterisk? Can i use kamailio for sip trunks to asterisk and carry rtp and natted clients media streams rather than register to asterisk? Many thanks, Taff..

3 9

[Kamailio-Users] default config file
by Elena-Ramona Modroiu 26 Feb '09

26 Feb '09

Hi, looking at the default config file, we deliver something pretty useless and very insecure. I propose to add to it several features so it becomes really usable: - mysql support - even some use radius for aaa, mysql is for sure used for usrloc. Those using other db drivers can just replace the module - auth against db and check auth username against from/to username to avoid spoofing - accounting - at least to syslog to store more than default data which gives nothing useful right now. Thinking to from/ruri username and domain at least in order to have caller and callee details - usrloc to database - nat traversal support with nathelper and rtpproxy - if rtpproxy is not running, will be just warnings at startup - the first part of the config will have some help of how to configure the environment to start the sip server (e.g., add user, ...) What do you think? Regards, Ramona

5 7

[Serusers] Opensips--Ser
by Parikhita Behera 26 Feb '09

26 Feb '09

Dear Sir I am new to SER . So I want to know , is Opensips is Open source?. Does it need license?. What operating system needed to install the SER. Does SER support totally telephony, pc-pc , pc-phone, calling card , call back , IPPBX etc.. How many systems needed to install to work the complete solution. If possible , you can give me the installation and configuration ducumentation so that I can try to do the complete solution. Waiting for your reply. Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

2 1

[Serusers] Need easiest way to set/clear a flag and refer to it later in same call session
by Frank Durda IV 26 Feb '09

26 Feb '09

I need a simple way to set either a boolean flag (or integer value) during the INVITE of a given call in ser.cfg, and test that value via "if" elsehere in ser.cfg during the 183/200 response messages for that same call. This would be a data object unique to a given call, which should disappear/lose persistence/end scope when a given call ends. Not global, no need to write the value to disk, doesn't need to survive a restart of SER, etc. The various SER documents describe a number of things that could do this, but most seem overly complex, storing the object in a MYSQL database on disk or something, and these all seem so excessive. There also appear to be a few built-in flags, but the documentation suggests that all are actually there for specific uses and so use for other things might not work. I simply want to make a determination of something during the INVITE for call 12345, and be able to refer to that flag or integer value later on when the various response messages for the same call (#12345) are coming back through. I can't make the same determination on each message after the INVITE because the original src_ip address value where the INVITE came from doesn't seem to be in a testable variable when handling reply messages, or at least not in one that is clearly documented. I've chased a number of possible ways this might be done, but tire of being referred away from the doc that came with the source and over to web sites that tell me this or that module that might do part of what I need is deprecated and to use this other module, but the web site for the other module says that the first module is better suited to what I want to do (a circle), doesn't show how the set value can actually be tested, or the web page just doesn't work anymore. So, assuming there is a simple way to do this, I need a quick code snip showing how to set/clear/assign a value to such a beast within INVITE handling, and a quick code snip of how to test it in an "if" statement during 183/200/110 type response messages. Whatever it is, the value set, cleared or being tested in one call should not affect any other call. Thanks in advance!

2 1

[Serusers] Opensips--Ser
by Parikhita Behera 26 Feb '09

26 Feb '09

Dear Sir I am new to SER . So I want to know , is Opensips is Open source?. Does it need license?.. What operating system needed to install the SER. Does SER support totally telephony, pc-pc , pc-phone, calling card , call back , IPPBX etc.. How many systems needed to install to work the complete solution. If possible , you can give me the installation and configuration ducumentation so that I can try to do the complete solution. Waiting for your reply. Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

1 0

[Serusers] How to set the query into database and get answers with exec_msg?
by 邱磊 26 Feb '09

26 Feb '09

Hi ,I am a student studying in VoIP and implementing my theory in SER. I try to access mysql database to get some information, but have a problem aboout paramaters of exe_msg. My goal is :count the number of contact addresses that certain user registered, and take this is my solution: exec_msg(' Count="select count(contact) from location where username=\"$SIP_HF_CONTACT\" "' CountResult=`mysql -user -pheslo -e "$Count" ser`; ') What is the result of "CountResult" ? I suppose originally that it will be the the number of the query, but it seems not......... and i try the exec_msg('rm -r test'); but it failed; Dose the function wored? and how to use it,Hhw can I get my answer what I want to get ? Hoping someone can give me some advises ^^. Thannks in advance. 2009-02-26 邱磊

1 0

[Kamailio-Users] Handling CANCEL and 487
by Juan Perez 25 Feb '09

25 Feb '09

Hello I am having a prob handling the CANCEL in this scenario Kamailio 1.4 doing basic load balancing with LCR module. Calling party makes a call, sends it to Kamailio, Kamailio looks up the destination gw and sends the call to the dest IP. I am using statefull mode and loose routing. The calling party makes the call and while it is ringing it hangs up. Kamailio receives the CANCEL and forward it to the dest IP, it receives the 487 from dest IP and pass it back to the calling party. This is when it gets weird. The calling party sends the ACK to the 487 and it looks like the proxy [kamailio] ignores it and keep sending the 487 to the calling party. Calling party IP: 192.168.1.200 kamailio[proxy]: 192.168.1.50 dest IP: 192.168.11 PLease see below a tethereal output and my config file. I would appreciate any help to help me fix this prob. [root@proxy1 ~]# tethereal -i eth1 -S -w /root/call-with-cancel3.pcap udp port 5060 Running as user "root" and group "root". This could be dangerous. Capturing on eth1 0.000000 192.168.1.200 -> 192.168.1.50 SIP/SDP Request: INVITE sip:664050495179280@192.168.1.50, with session description 0.000702 192.168.1.50 -> 192.168.1.200 SIP Status: 100 Giving a try 0.000774 192.168.1.50 -> 192.168.1.11 SIP/SDP Request: INVITE sip:664050495179280@192.168.1.11:5060;transport=udp, with session description 0.044219 192.168.1.11 -> 192.168.1.50 SIP Status: 100 Trying 0.731527 192.168.1.11 -> 192.168.1.50 SIP/SDP Status: 183 Session Progress, with session description 0.731585 192.168.1.50 -> 192.168.1.200 SIP/SDP Status: 183 Session Progress, with session description 1.869161 192.168.1.200 -> 192.168.1.50 SIP Request: CANCEL sip:664050495179280@192.168.1.50 1.869626 192.168.1.50 -> 192.168.1.200 SIP Status: 200 canceling 1.869738 192.168.1.50 -> 192.168.1.11 SIP Request: CANCEL sip:664050495179280@192.168.1.11:5060;transport=udp 1.869942 192.168.1.11 -> 192.168.1.50 SIP Status: 487 Request Terminated 1.869946 192.168.1.11 -> 192.168.1.50 SIP Status: 200 OK 1.870243 192.168.1.50 -> 192.168.1.11 SIP Request: ACK sip:664050495179280@192.168.1.11:5060;transport=udp 1.870273 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 1.871112 192.168.1.200 -> 192.168.1.50 SIP Request: ACK sip:664050495179280@192.168.1.50 2.317041 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 3.317025 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 5.317028 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 9.317036 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 13.317043 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 17.317049 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 21.317057 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 25.317064 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 29.317071 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 23 packets captured This is my routing logic: ####### Routing Logic ######## # main routing logic route{ xlog("L_INFO", "mylog: starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","mylog: Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_INFO","mylog: Message too big.\n"); exit; }; if (!method=="REGISTER") { record_route(); xlog("L_INFO","mylog: Recording Route. Method: [$rm].\n"); } if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); route(1); }; if (method=="INVITE") { if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } if(!next_gw()){ sl_send_reply("503", "Unable to find a gateway"); exit; } route(1); } #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } } route[1] { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); if (!t_relay()) { xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); sl_reply_error(); }; exit; }

2 1

Re: [Kamailio-Users] Need to handle an upstream carrier error - Duplicate INVITE Messages
by Geoffrey Mina 25 Feb '09

25 Feb '09

Hello, I have a carrier who provides PSTN gateway services. They have multiple redundant sip gateway devices in their network. The problem occurs when one of their devices starts to have issues. I will receive an INVITE request from both gateways with the same call-id. The problem is that my Kamailio system doesn't detect that I already set a call up for the INVITE once, and forwards the request to another server in the dispatcher list. What I end up with is a call on two asterisk servers, but only one has the actual RTP stream. The BYE request gets routed to the wrong server, and everything just gets screwy. If anyone could provide any hint on how I might be able to deal with this scenario, I would really appreciate it. I have attached my current config file, and the following is a link to a google spreadsheet which shows the SIP trace. http://spreadsheets.google.com/ccc?key=pU5i2J6Ck3b519-_M6Et3cw I have masked my IP addresses for my own sanity. XX.XX.XX.179 - Kamailio SIP Gateway XX.XX.XX.189 - Asterisk1 XX.XX.XX.186 - Asterisk2 Thanks! Geoff

2 5

[Kamailio-Users] openser not distributing to all DNS SRV entries
by Brian 25 Feb '09

25 Feb '09

Hi, I have a SRV setup with three equal priority entries and equal weights. I expected openser (version 1.3.3) to distribute the INVITE requests equally to all 3 hosts, but it seems that all requests are forwarded to only one host. The SRV entries look like this: _sip._udp.mydomain.com. 8805 IN SRV 100 33 5060 proxy1.mydomain.com. _sip._udp.mydomain.com. 8805 IN SRV 100 34 5060 proxy1.mydomain.com. _sip._udp.mydomain.com. 8805 IN SRV 100 33 5060 proxy1.mydomain.com. And my routing for INVITE request looks like this: rewritehost("mydomain.com"); rewriteport(""); if (!t_relay()) { sl_reply_error(); }; exit; Am I wrong in expecting this to distribute equally to the three proxy hosts? I thought this would just "work". Thanks, Brian.

3 3

Re: [Kamailio-Users] event in kamailio
by Marion.DELEY＠fr.thalesgroup.com 25 Feb '09

25 Feb '09

Hi, Finally, I've used the send() function in kamailio.cfg with the @ of my UDP server to know when invite and register requests arrive. Thanks for your help. marion -----Message d'origine----- De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Envoyé : mardi 24 février 2009 21:32 À : Marion.DELEY(a)fr.thalesgroup.com Cc : users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] event in kamailio Marion.DELEY(a)fr.thalesgroup.com wrote: > Hi everybody, > > I'm novice in using Kamailio and I need to know if it's possible to create > event from kamailio.cfg to another program. > Actually, I use MI command but I would find a way to force kamailio to send > me messages and not just responses. If the other programm supports SIP or HTTP you can send SIP requests from kamailio.cfg and generate HTTP requests. regards klaus

2 1

[Kamailio-Users] event in kamailio
by Marion.DELEY＠fr.thalesgroup.com 24 Feb '09

24 Feb '09

Hi everybody, I'm novice in using Kamailio and I need to know if it's possible to create event from kamailio.cfg to another program. Actually, I use MI command but I would find a way to force kamailio to send me messages and not just responses. Some help? thanks marion

3 2

[Kamailio-Users] Error compiling CARRIERROUTE module
by Asim Riaz 24 Feb '09

24 Feb '09

Hello, I am trying to compile CARRIERROUTE module, I am getting errors below; Kamailio 1.4 make prefix=/usr modules=modules/carrierroute modules make[1]: Entering directory `/usr/local/src/kamailio/sip-server/modules/carrierroute' Compiling carrierroute.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"carrierroute"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DSVNREVISION='"2:5452M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c carrierroute.c -o carrierroute.o In file included from carrierroute.c:30: ../../sr_module.h:48:19: error: dlfcn.h: No such file or directory In file included from /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/syslimits.h:7, from /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/limits.h:11, from ../../parser/../flags.h:32, from ../../parser/msg_parser.h:46, from ../../sr_module.h:50, from carrierroute.c:30: /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/limits.h:122:61: error: limits.h: No such file or directory In file included from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: ../../parser/../ip_addr.h:38:20: error: string.h: No such file or directory ../../parser/../ip_addr.h:39:23: error: sys/types.h: No such file or directory ../../parser/../ip_addr.h:40:24: error: sys/socket.h: No such file or directory ../../parser/../ip_addr.h:41:24: error: netinet/in.h: No such file or directory ../../parser/../ip_addr.h:42:19: error: netdb.h: No such file or directory In file included from ../../parser/../ip_addr.h:46, from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: ../../parser/../dprint.h:54:20: error: syslog.h: No such file or directory ../../parser/../dprint.h:55:18: error: time.h: No such file or directory In file included from ../../parser/../ip_addr.h:46, from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: any idea ? Thanks in advance. Asim

1 1

Re: [Kamailio-Users] help me in ldap kamailio
by Henning Westerholt 24 Feb '09

24 Feb '09

On Tuesday 24 February 2009, Husna Bte Mad Baguri wrote: > There is no error msg.just there is no file name ldap.cfg in > usr/local/etc/kamailio.lets me explain how i install. > > 1.i using ubuntu 8.10 > 2.i use kamailio 1.4.3 with tls > 3.i install open ldap --> libldap2-dev,ldap-utils,slapd > 4.then i exclude module =ldap Hi Husna, did i understand you correctly, you excluded the ldap module from compilation? This way it will not be compiled. > 5.then i install.no error but no file ldap.cfg > 6.can you explain what next. It seems that the ldap.cfg file is not installed per default. You find this file in modules/ldap/etc/ in the kamailio 1.4.3 directory. Just copy it to the location you want, and probably adapt it to your setup. Cheers, Henning

1 0

[Kamailio-Users] How can I use domain module?
by Ramu 24 Feb '09

24 Feb '09

Hi, I would like to use domain module to support multiple domain users. How can move forward on this? Thanks, Ramu

5 8

[Kamailio-Users] help me in ldap kamailio
by Husna Bte Mad Baguri 24 Feb '09

24 Feb '09

Can someone help me,i counter this error when try to install ldap in kamailio Compiling ldap_api_fn.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"ldap"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c ldap_api_fn.c -o ldap_api_fn.o ldap_api_fn.c:38:18: error: ldap.h: No such file or directory In file included from ldap_api_fn.c:40: Plzzzzz help me regards Husna This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

2 1

[Kamailio-Users] help me in ldap kamailio
by Husna Bte Mad Baguri 24 Feb '09

24 Feb '09

Can someone help me,i counter this error when try to install ldap in kamailio Compiling ldap_api_fn.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"ldap"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c ldap_api_fn.c -o ldap_api_fn.o ldap_api_fn.c:38:18: error: ldap.h: No such file or directory In file included from ldap_api_fn.c:40: Plzzzzz help me regards Husna This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

1 1

[Serusers] Cleaning bad npdi=yes and rn=VALUE tags from INVITEs
by Frank Durda IV 24 Feb '09

24 Feb '09

I have a client who occasionally sends calls where there is a npdi=yes tag (and there may or may not be a rn=VALUE tag present with it), BUT the answer these two tags are providing is wrong! That is, the number in question has or hasn't been ported, but the information provided in the INVITE message doesn't match reality. Either a rn=VALUE should have had value X and it instead has value Y, OR there is no rn=VALUE at all, and there should have been one, OR the number hasn't actually been ported, and no rn=VALUE should have been sent at all but we got a junk one, that probably isn't the right LRN for the true destination of the call. Because of these wrong npdi=yes tags, our PSTN gateway switch ends up failing these incorrectly-marked call, because it takes the provided rn=VALUE or lack of one to be correct and so it doesn't do its own LNP query, and the one provided in the SIP INVITE is sometimes just flat out wrong. If the PSTN gateway switch had ignored these suspect npdi=yes and rn=VALUE tages and simply done the LNP query itself, in virtually all cases these otherwise doomed calls would have completed successfully. For this particular customer, the problem of bogus values appears to be a permanent artifact. They either don't know how to fix it, or don't think it is important, of they are spoofing it to avoid being billed for LNP queries. Whatever the reason, it causes trouble tickets. So, we would like to have SER strip the npdi=yes and any rn=VALUE tags from the INVITE when it is this customer (known by src ip), so that the PSTN switch will do the LNP query itself on the SS7 side and get the call routing right. (We have already determined that getting the outfit that is sending the junk to do the right thing isn't going to happen any time soon.) Does anybody have any suggestions as how to write the ser.cfg code to edit-out a npdi=yes and rn=VALUE if they are present in INVITEs? We just want to always throw npdi=yes and any rn=VALUE that shows up if the call is coming from this given customer, basically treating all npdi=yes and rn= tags coming from this customer as being untrustworthy. There has to be an easier way than stripping the thing down and reassembling a new one with repeated strip(1) type operations, which looks to be extremely inefficient for this task and probably really meant only for stripping country codes and such off the phone numbers, not for cleaning up bad tags. Thanks, in advance!

1 0

Re: [Kamailio-Users] LCR in Kamailio 1.4
by Ovidiu Sas 23 Feb '09

23 Feb '09

Make sure that from_uri is NULL and not a white space. Please copy the mailing list in your reply. Regards, Ovidiu Sas On Thu, Feb 19, 2009 at 11:25 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: > > > On Thu, Feb 19, 2009 at 4:16 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >> Is your R-URI starting with a '0'? >> You have defined a single route for all '0' prefix destinations. > > yes R-URI is starting with 0. >> >> >> Regards, >> Ovidiu Sas >> >> On Thu, Feb 19, 2009 at 11:06 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: >> > >> > >> > On Thu, Feb 19, 2009 at 3:53 PM, Daniel-Constantin Mierla >> > <miconda(a)gmail.com> wrote: >> >> >> >> Hello, >> >> >> >> On 02/19/2009 05:34 PM, Asim Riaz wrote: >> >>> >> >>> Hello, >> >>> I am using LCR, for some reason load_gws() function is not finding the >> >>> gateway in DB, it was working in the start then added some more routes >> >>> and >> >>> after that everything stopped working, I am not sure where to start >> >>> troubleshooting, any idea ? >> >> >> >> run kamailio with higher debug mode (e.g., debug=5) in configuration >> >> file. >> >> >> >> You should get more debug messages in the syslog, that may give you >> >> some >> >> hints. look for lines containing 'lcr'. >> >> >> >> Cheers, >> >> Daniel >> > >> > Hi Daniel, >> > I tried higher debug level, not getting anything related to LCR all i >> > am >> > getting XLOG messages "unable to find Gateway". >> > >> > Thanks, >> > >> > Asim >> > >> >> >> >>> >> >>> kamctl lcr show >> >>> >> >>> lcr routes >> >>> +----+--------+----------+--------+----------+ >> >>> | id | prefix | from_uri | grp_id | priority | >> >>> +----+--------+----------+--------+----------+ >> >>> | 9 | 0 | | 1 | 1 | >> >>> +----+--------+----------+--------+----------+ >> >>> lcr gateways >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | gw_name | ip_addr | port | uri_scheme | transport | grp_id >> >>> | >> >>> strip | tag | flags | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | XXXX | XXXX | 5060 | 1 | 1 | 1 | >> >>> 0 >> >>> | | 0 | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> >> >>> >> >>> >> >>> my routing script >> >>> >> >>> if(!load_gws()) { >> >>> xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); >> >>> sl_send_reply("503", "unable to find gateway"); >> >>> exit; >> >>> } >> >>> Thanks in advance. >> >>> >> >>> Asim >> >>> >> >>> ------------------------------------------------------------------------ >> >>> >> >>> _______________________________________________ >> >>> Kamailio (OpenSER) - Users mailing list >> >>> Users(a)lists.kamailio.org >> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> >> Daniel-Constantin Mierla >> >> http://www.asipto.com >> >> >> > >> > >> > _______________________________________________ >> > Kamailio (OpenSER) - Users mailing list >> > Users(a)lists.kamailio.org >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > > >

4 15

[Kamailio-Users] problem with 200ok
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, I have Asterisk1---SER---Asterisk2. When I do INVITE from the left, --the asterisk2 send 200ok to the SER --the SER forward to the Asterisk1 --but the asterisk1 directly send the ACK to Asterisk2 Asterisk2 retransmit the 200ok… and error. I think that it need that the ACK come from the SER and not directly from the Asterisk1. So, how can I detect a 200ok and reply a ACK with my SER? Or, anyone have ever seen that problem? Thank you Next: my capture problem SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv 6§¢Iœÿ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 200 OK v: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.0 v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@192.168.1.82:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@IP_PHONE:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Via: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.2 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport=5060 Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 ACK sip:URI_TEST@IP_ASTERISK SIP/2.0 Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK7a6d7cf4;rport From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Contact: <sip:170725014@IP_PROVIDER_ASTERISK> Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 ACK User-Agent: MARS Max-Forwards: 70 Content-Length: 0 Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 2

[Kamailio-Users] NAT traversal
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, Is it possible to do NAT traversal without RTP-PROXY ? If yes, have you an example? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

7 18

[Kamailio-Users] how to prepend area code into a local call phone number
by Jinsong Hu 23 Feb '09

23 Feb '09

Hi, Sir: In the US, assume a person from 1-408-1234567 is dialing phone number 1234568. how do I figure out the caller's country code and area code and prepend to the local number ? I tried prefix($(from.user{s.substr, 0, 4})); and openser reported as parser error in config file, bad argument, string expected. Jimmy

2 1

[Kamailio-Users] Using set_advertised_address with variable
by catalina oancea 23 Feb '09

23 Feb '09

Hi Does anybody know if there is a way to use the set_advertised_address function with an avp or variable parameter instead of plain string? I need to change the IP in the Via header with the value of a variable. Thanks Catalina

2 1

[Kamailio-Users] missing file ldap.h
by Husna Bte Mad Baguri 23 Feb '09

23 Feb '09

Dear Whom May Concern, When I try to compile ldap module in kamailio I get error,ldap.h :No such file or directory. Where can I get that file.please someone help me. Regards Husna ________________________________ This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

2 1

Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 20 Feb '09

20 Feb '09

Hello, On 02/13/2009 12:08 PM, Abdul Hakeem wrote: > Hi, > Are you able to put me in touch with the guys using the SDP parser ? > direct your emails to mailing list all the time. Those people are there, the developer of SDP parser and others using it. You will get much faster response and maybe better than writing private emails. Thanks, Daniel > Cheers, > Abdul Hakeem > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: 11 February 2009 15:20 > To: alhakeem(a)ipextelecom.net; kamailio > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > >> Hi, >> >> Just wanted to touch base on the parser to find out if you are still >> pursuing this. >> >> > it is still on my todo list, however didn't make it in the upcoming 1.5.0. > Ovidiu Sas did some commits to the SDP parser and I know couple of others > using it with success. > > Please open a feature request at: > http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse > > It is easier to track and not forgot about it for the next release. > > Thanks, > Daniel > > > >> I am in the mind of purchasing some Cisco equipment to do the Sip-MAP >> gateway from a GSM network. >> I have someone else from the OpenBTS with me on this. >> The object is to provide a full fledged sip based routing of voice, >> sms and multimedia calls from gsm handsets to BTS with a Sip server >> > backend. > >> Cheers, >> Abdul Hakeem >> >> -----Original Message----- >> From: users-bounces(a)lists.kamailio.org >> [mailto:users-bounces@lists.kamailio.org] On Behalf Of >> Daniel-Constantin Mierla >> Sent: 03 September 2008 17:45 >> To: Kristian Kielhofner >> Cc: users(a)lists.kamailio.org >> Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc >> >> Hello, >> >> On 09/03/08 19:10, Kristian Kielhofner wrote: >> >> >>> Hello everyone, >>> >>> Feel free to set $OSSPROXY to whatever you like = >>> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >>> >>> $OSSPROXY is my favorite SIP toolkit. In addition to being an >>> excellent proxy and registrar it serves as a platform to do SIP >>> mangling and wrangling. With that in mind let me propose a crazy >>> idea... >>> >>> How hard would it be to get $OSSPROXY to be able to parse what is >>> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >>> no business getting too involved in the SDP but I feel it makes sense >>> in some instances (like this one). What if I want to make a routing >>> decision based on an encapsulated ISUP parameter of some sort? Who >>> knows what it could be!?!? Maybe there are some other uses but that >>> makes the most sense and seems to be the most practical. It should >>> be possible to develop an $OSSPROXY module to: >>> >>> - Use the SDP parser to find the ISUP SDP part: >>> >>> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >>> >>> - Pass that part into a "real" ISUP parser (if necessary). I >>> couldn't seem to find the perfect library but perhaps something like >>> this could >>> work: >>> >>> http://www.openss7.org/isup.html >>> >>> - Make the ISUP parameters available via AVPs, pseudo variables, etc >>> for use in the $OSSPROXY script. >>> >>> I don't really have a need for this but it seems like it could be >>> cool. Am I nuts? >>> >>> >>> >>> >> I would expose the body as a dynamic structure, so components will be >> accessible in the script, e.g. (naming schema just for example), >> >> if($body(0=>content-type)=="application/sdp") >> { >> if($body(0=>m=>type) == "audio") >> { >> } >> } >> >> Based on the available content type parsers (sdp, isup), the structure >> has different attributes. >> >> Cheers, >> Daniel >> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

5 13

[Kamailio-Users] lookup("location") problem
by BERGANZ François 19 Feb '09

19 Feb '09

Hello, I do as the example in the module registrar doc: lookup("location"); switch ($?) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; And SER report me ERROR:core:pv_parse_spec: wrong char [$/36] in [$?] at [0 (0)] Know you why? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 3

[Kamailio-Users] call keepalived using rtimer?
by Noel R. Morais 19 Feb '09

19 Feb '09

Hi Guys, I would like to know if is possible to make a keepalive for active calls (dialog module)? Well, I think it is but is there a better way than making a selec into the dialog table and building the INVITEs? Thanks, Noel

6 8

[Kamailio-Users] SIP URI Extracting
by Ramu 19 Feb '09

19 Feb '09

Hi, I would like to extract uri part from SIP Request uri. Say I received a INVITE sip:1234567890@ip Now I would like to take 4567890 the uri and send it as a new INVITE. Ramu

3 2

[Kamailio-Users] LCR in Kamailio 1.4
by Asim Riaz 19 Feb '09

19 Feb '09

Hello, I am using LCR, for some reason load_gws() function is not finding the gateway in DB, it was working in the start then added some more routes and after that everything stopped working, I am not sure where to start troubleshooting, any idea ? kamctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 9 | 0 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateways +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | tag | flags | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | XXXX | XXXX | 5060 | 1 | 1 | 1 | 0 | | 0 | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ my routing script if(!load_gws()) { xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); sl_send_reply("503", "unable to find gateway"); exit; } Thanks in advance. Asim

3 6

[Kamailio-Users] related free voip providers
by Daniel-Constantin Mierla 19 Feb '09

19 Feb '09

Hello, I posted a summary of three free voip providers running sip-router.org related sip servers. It is based on my experience as user of all these (an other) services. The providers listed on the page are not ranked, just presented in alphabetic order, pretty summary about each one. I plan a longer version separately, one by one. It is a good starting point for those willing to see what can be done with Kamailio (OpenSER) and/or SIP Express Router (SER). Another benefit is for users looking to get an independent SIP address, not related to current job or location. I created the post after some private inquires, but I think it worth for public audience: http://miconda.wordpress.com/2009/02/17/top-three-free-voip-services/ In the same spirit, you can check the second edition of (my personal) Kamailio (OpenSER) awards: http://miconda.wordpress.com/2009/01/19/kamailio-openser-2008-awards/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] SIREMIS 0.9.2 released - Support for Carrierroute
by Elena-Ramona Modroiu 19 Feb '09

19 Feb '09

Hi all, a new version of SIREMIS, web management interface for Kamailio (OpenSER) and SIP-Router.org, is out as 0.9.2. Main improvements refer to addition of support to manage the carrierroute module: * management of carrierroute table * management of carrierfailureroute table * management of carrier_name table * management of carrier_domain table View screenshots: http://www.asipto.com/gallery/v/siremis/ Demo site (it works on a database with random data, username: admin, password: admin): http://siremis.asipto.com/demo/ Download: http://siremis.asipto.com/pub/downloads/siremis/ Install/Upgrade documentation: http://siremis.asipto.com/install/ Regards, Ramona

1 0

[Kamailio-Users] Config
by NYam 19 Feb '09

19 Feb '09

Dear Sir, I have problem . wenn i root:~ # kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root:~ # kamailio -c Feb 17 11:01:49 [10767] ERROR:core:main: loading config file(/usr//etc/kamailio/kamailio.cfg): No such file or directory Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called root:~ # kamailio -f /etc/kamailio/kamailio.cfg then i see udp: 127.0.0.1 [127.0.0.1]:5060 ********************* ********************** tcp: 127.0.0.1 [127.0.0.1]:5060 ok Where is Problem ? can you help me? with kind regards Nyam

3 2

[Kamailio-Users] Kamailio 1.4.3 Carrierroute BYE does not
by unknown＠1.org 19 Feb '09

19 Feb '09

Hello: I have this scenario: I already rad about the bye2bye, the dialog module, i was not able to see any light there. Here is the scenario A--GW--------Kamailio(carrierroute)---------GW1---B |--------------GW2 |--------------GW3 (etc) the capture |<-------------BYE---- -<----NOTHING HERE<<<-| the By does go back if the B user hangs it is not even attempted in the kamailio, per the traces. the BY generates records in the CRD from B side if i hang manually the bye generates CDR records from the A side. Again the Bye is understood by the kamailio and generates the CDR records, that is Not the problem, the Problem is i have the user A in silence, i am talking about the Signaling Only, not Media, no NAT. I even tried the mediaproxy (just trying) the config is like the: http://voipembedded.com/resources/openser_cr.cfg i added if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails exit; } route(10); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(200); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); xlog("L_WARN", "[$mi] discarding ACK\n"); exit; } this is the main difference for now, i changed to use as a basis example the Example 1.45. Configuration example - Routing to user tree. same results. i tried adding the record_route in the invite also, same result. At this point i have no clue what i am missing. Please let me know if this should work or not, and if yes, based in that basic example, what i should add.

2 8

[Kamailio-Users] A2billing Project <-> Openser
by BERGANZ François 18 Feb '09

18 Feb '09

Hello, I use Asterisk2billing to do prepaidit works fine! But, now, I need to auth my users in the openser and forward the call by loadbal to asterisk(s) which use Asterisk2billing. Anybody know how to do? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 14

[Kamailio-Users] SIP-T
by Kaeman Chris 18 Feb '09

18 Feb '09

Hello, Does opensips support SIP-T? Thank you! KC Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 3

[Kamailio-Users] SEMS (bugfix) release 1.0.1
by Stefan Sayer 17 Feb '09

17 Feb '09

Hello, The 1.0.1 release of SEMS, the SIP Express Media Server [0], is now available for download [1]. The 1.0.1 release contains all fixes applied to the 1.0 branch, and should be a drop-in replacement for anyone still using 1.0.0. Update is recommended to everyone who is still using 1.0.0-rcX or any later version from the 1.0 branch. Debian packages for etch/64 are also available [2]. Best Stefan Sayer [0] http://iptel.org/sems/ [1] http://ftp.iptel.org/pub/sems/sems-1.0.1.tar.gz [2] http://ftp.iptel.org/pub/sems/1.0/1.0.1/packages/debian Apologies if you receive multiple copies of this message. -- 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

1 0

[Kamailio-Users] minutes for IRC meeting 17.02.2009, release dates
by Henning Westerholt 17 Feb '09

17 Feb '09

Hi all, the minutes for todays IRC meeting are now available in our wiki [1]. Among the discussed topics we decided about a date for the upcoming 1.5.0 release, this is scheduled for the 2th march 2009. The first release that will use the new common sip-router core [2] is scheduled for May. After 1.5.0 we will also do a bug fix release (1.4.4) for our latest stable branch. All dates can be found in our wiki as well at [3]. Cheers, Henning [1]http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-minutes… [2]http://sip-router.org/benefits/ [3]http://www.kamailio.org/dokuwiki/doku.php/development:release-policy#date… -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] IRC meeting 17th februar, 15:00 UTC
by Henning Westerholt 17 Feb '09

17 Feb '09

Hi all, as the release cycle for 1.5.0 slowly come to an end, i think its time for a new IRC developer meeting, to focus on the upcoming development. I propose next Tuesday, 17th Feb 15:00 UTC. I've created a wiki page to collect an agenda for this meeting, and already added some content i would like to discuss. Please add your yours also and comment on already existing topics. Agenda: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-17… IRC meetings page: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meetings Cheers, Henning

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[Serusers] Problem with REGISTER
by buslik 17 Feb '09

17 Feb '09

Hi all, I have just default ser.cfg And it seems that if (uri==myself) never happens. SER trying to relay to <my domain name>:5060 and there my Gigaset C470IP just replies 405 Not Allowed. How to explain to the SER that it has <my domain name>? I have put alias=<my domain name> into my ser.cfg and SIP_DOMAIN=<my domain name> into env. It does no help. How to get any log out of it? -- Slava.

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[Kamailio-Users] Lost
by daniel medina 17 Feb '09

17 Feb '09

Hi everybody, I am trying to set up OpenSer as a Load balancing for Asterisk. I really need to know where to start. Please can anyone give me a hint where to star or if there is any tutorial??? Thanks in advance Daniel

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Re: [Kamailio-Users] ERROR:uac:replace_from
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

Are you sure uac_replace_from() was called for initial invite? There has to be a new param (vsf) in the Recor-Route header and it is not there. Cheers, Daniel On 02/16/2009 04:50 PM, BERGANZ François wrote: > No. Time Source Destination Protocol Info > > [...] > > There is no 200OK because, the bye isn't good! > > > Thank you > > > Cordialement, > BERGANZ François > > > http://www.acropolistelecom.net >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Envoyé : lundi 16 février 2009 15:37 > À : BERGANZ François > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] ERROR:uac:replace_from > > > > On 02/16/2009 04:07 PM, BERGANZ François wrote: > >> Now, I have modparam("uac","from_restore_mode","none") >> >> I control everything! Nothing in automatic! >> >> > :-) -- the power in your hands -- as said in the previous email, if you > send me the invite and appropriate 200ok, I can tell whose fault is. > > Cheers, > Daniel > > >> Cordialement, >> BERGANZ François >> >> >> http://www.acropolistelecom.net >>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >> >> -----Message d'origine----- >> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Envoyé : lundi 16 février 2009 14:07 >> À : BERGANZ François >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >> >> >> >> On 02/16/2009 03:03 PM, BERGANZ François wrote: >> >> >>> Then, if it is automatic: >>> >>> If I change with uac_replace_from() the INVITE, the bye have to be the same ? >>> In my case, it doesn't >>> >>> >>> >> Paste the BYE here, maybe the Route parameter used for auto-update is >> missing... >> >> Cheers, >> Daniel >> >> >> >>> Cordialement, >>> BERGANZ François >>> >>> >>> http://www.acropolistelecom.net >>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>> >>> >>> -----Message d'origine----- >>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Envoyé : lundi 16 février 2009 12:39 >>> À : BERGANZ François >>> Cc : users(a)lists.kamailio.org >>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>> >>> >>> >>> On 02/16/2009 01:33 PM, BERGANZ François wrote: >>> >>> >>> >>>> It is for a bye request >>>> >>>> >>>> >>>> >>> BYE is withing dialog (practically ending the dialog). You do not need >>> to replace the From for it. It is done automatically -- see my other post. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >>>> Cordialement, >>>> BERGANZ François >>>> >>>> >>>> http://www.acropolistelecom.net >>>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>>> >>>> >>>> -----Message d'origine----- >>>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>>> Envoyé : lundi 16 février 2009 12:25 >>>> À : BERGANZ François >>>> Cc : users(a)lists.kamailio.org >>>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>>> >>>> Hello, >>>> >>>> On 02/16/2009 01:10 PM, BERGANZ François wrote: >>>> >>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> I have an error with uac_replace_from() function. >>>>> >>>>> I want to do uac_replace_from(“$fn”,” sip:181812012@217.64.x.x”); >>>>> >>>>> I have that error: ERROR:uac:replace_from: decline FROM replacing in >>>>> sequential request in auto mode (has TO tag) >>>>> >>>>> Have you an idea? >>>>> >>>>> >>>>> >>>>> >>>>> >>>> you call uac_replace_from() for SIP messages within dialog and the >>>> replace mode is auto (which means is done automatically by kamailio) - >>>> therefore will be a duplicate. >>>> >>>> Either check for to-tag when you do the replacement or set replace mode >>>> to manual. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] ERROR:uac:replace_from
by BERGANZ François 16 Feb '09

16 Feb '09

Hello, I have an error with uac_replace_from() function. I want to do uac_replace_from($fn, sip:181812012@217.64.x.x); I have that error: ERROR:uac:replace_from: decline FROM replacing in sequential request in auto mode (has TO tag) Have you an idea? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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Re: [Kamailio-Users] ERROR:uac:replace_from
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

On 02/16/2009 03:36 PM, BERGANZ François wrote: > the bye receive by the SER > The route parameter is missing, please send me the INVITE going out from kamailio and the 200ok for invite received by Kamailio. Should reveal whose fault is. Cheers, Daniel > thank you > > > > [...] > > > > Cordialement, > BERGANZ François > > > http://www.acropolistelecom.net >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Envoyé : lundi 16 février 2009 14:07 > À : BERGANZ François > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] ERROR:uac:replace_from > > > > On 02/16/2009 03:03 PM, BERGANZ François wrote: > >> Then, if it is automatic: >> >> If I change with uac_replace_from() the INVITE, the bye have to be the same ? >> In my case, it doesn't >> >> > > Paste the BYE here, maybe the Route parameter used for auto-update is > missing... > > Cheers, > Daniel > > >> Cordialement, >> BERGANZ François >> >> >> http://www.acropolistelecom.net >>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >> >> >> -----Message d'origine----- >> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Envoyé : lundi 16 février 2009 12:39 >> À : BERGANZ François >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >> >> >> >> On 02/16/2009 01:33 PM, BERGANZ François wrote: >> >> >>> It is for a bye request >>> >>> >>> >> BYE is withing dialog (practically ending the dialog). You do not need >> to replace the From for it. It is done automatically -- see my other post. >> >> Cheers, >> Daniel >> >> >> >>> Cordialement, >>> BERGANZ François >>> >>> >>> http://www.acropolistelecom.net >>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>> >>> >>> -----Message d'origine----- >>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Envoyé : lundi 16 février 2009 12:25 >>> À : BERGANZ François >>> Cc : users(a)lists.kamailio.org >>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>> >>> Hello, >>> >>> On 02/16/2009 01:10 PM, BERGANZ François wrote: >>> >>> >>> >>>> Hello, >>>> >>>> I have an error with uac_replace_from() function. >>>> >>>> I want to do uac_replace_from(“$fn”,” sip:181812012@217.64.x.x”); >>>> >>>> I have that error: ERROR:uac:replace_from: decline FROM replacing in >>>> sequential request in auto mode (has TO tag) >>>> >>>> Have you an idea? >>>> >>>> >>>> >>>> >>> you call uac_replace_from() for SIP messages within dialog and the >>> replace mode is auto (which means is done automatically by kamailio) - >>> therefore will be a duplicate. >>> >>> Either check for to-tag when you do the replacement or set replace mode >>> to manual. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] cfg scripts bank - inside group dial
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

Hello, there were many useful hints and examples of how to solve specific needs or add new features just by config language. However, it is hard to search and find them back in the archive after a while. Therefore I created a new wiki page to collect them. If you see such example flowing on the mailing lists, take it and add it there or at least set the link to the mailing list post. Here is the page: http://www.kamailio.org/dokuwiki/doku.php/cfg-scripts-bank:main-index I added first example that adds inside group dial feature. You can group users by interest and they can call each other using short numbers instead of entire extension. Check it here: http://www.kamailio.org/dokuwiki/doku.php/cfg-scripts-bank:group-dial Practically is less than 10 lines, with loading additional modules and setting the parameters. Looking forward to your contributions there! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Serusers] Serusers Digest, Vol 70, Issue 4
by 郭灵犀 16 Feb '09

16 Feb '09

help 在2009-02-06，serusers-request(a)lists.iptel.org 写道： >Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers >or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > >You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Serusers digest..." > > >Today's Topics: > > 1. Re: configure SER and SEMS to work together (Stefan Sayer) > 2. preload_udomain error (caio) > 3. Re: preload_udomain error (solved) (caio) > 4. Re: Error loading "trusted" table (permissions.so) (caio) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 05 Feb 2009 13:25:00 +0100 >From: Stefan Sayer <stefan.sayer(a)iptego.com> >Subject: Re: [Serusers] configure SER and SEMS to work together >To: Martin Lane <martin.l.lane(a)gmail.com> >Cc: sems(a)iptel.org, Serusers(a)lists.iptel.org >Message-ID: <498ADA9C.4050302(a)iptego.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > > >o Martin Lane [02/04/09 20:17]: >> I am following the 'mini-Howto on how to configure SER and SEMS to work >> together' guide on iptel.org. So I have a separate directory for >as it says at the top of that howto, with a more recent version of SEMS >this is not required any more; especially if you are using 1.1.0 >version. Also note that there is the 1.1.0 release available >(http://ftp.iptel.org/pub/sems/sems-1.1.0.tar.gz). > >For SEMS 1.1.0 (recommended!), you can use the default configuration, >just configure 'application' and the server's IP address and port used >in sems.conf. Then just send the call to the SIP IP:port, e.g. using ser >with t_relay_to_udp("<sip_ip>","<sip_port>"); > >for completeness, the following comments: > >> integrating the SER and SEMS functionality. For development purposes is >> there a switch to change the path were SER looks for the modules? As >no, you need to change every loadmodule line. > >> well control which version of SEMS is being used? I've installed Sip >> Express Media Server (1.1.0-rc1 (i386/linux)) as well as the version >> from the howto for testing. >> >> My basic problem is I currently get a version mismatch error: >> >> 0(9590) ERROR: module version mismatch for >> /usr/local/lib/ser/modules/sl.so; core: ser 0.9.6-sems (i386/linux); >> module: ser 0.9.6 (i386/linux) >> > >Your specific error comes from that the module you are trying to load >was not compiled with the same version, e.g. you have one ser installed >into /usr/local and you have another one in a local directory. > >Regards >Stefan > >> The appropriate modules are located in under my ser-sems test dir and I >> have not been successful in redirecting SER to look there. >> >> Also, I have modified ser.cfg and sems.cfg for my system but I did not >> find a default ser_sems.cfg. Is this file necessary and where is the >> default location? >> >> Thanks for any help or advice. >> >> mll >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > >-- >Stefan Sayer >VoIP Services > >stefan.sayer(a)iptego.com >www.iptego.com > >IPTEGO GmbH >Am Borsigturm 40 >13507 Berlin >Germany > >Amtsgericht Charlottenburg, HRB 101010 >Geschaeftsfuehrer: Alexander Hoffmann > > >------------------------------ > >Message: 2 >Date: Thu, 05 Feb 2009 12:18:40 -0200 >From: caio <elcaio(a)gmail.com> >Subject: [Serusers] preload_udomain error >To: serusers(a)lists.iptel.org >Message-ID: <498AF540.4030102(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hi all, > >Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) > >Have this problem when I stop ser service, and then start it again (ser >-E -D).. > > 0(0) preload_udomain: ERROR: bad callid record in table aliases > 0(0) preload_udomain: ERROR: for username 579981, contact >sip:579981@mydomain.com > 0(0) preload_udomain: ERROR: skipping... > >And finally it dies with: > > 0(0) get_connection(): Connection not found in the pool > Fallo de segmentaci??n (core dumped) > >It happens with many records on aliases table and happens too with only >one record: > >mysql> select * from aliases\G >*************************** 1. row *************************** > username: 579981 > domain: 10.10.1.1 > contact: sip:579981@mydomain.com > received: NULL > expires: 2009-02-05 11:45:21 > q: 1.00 > callid: >The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything > cseq: 42 >last_modified: 2009-02-05 11:45:46 > replicate: 0 > state: 0 > flags: 128 > user_agent: SIP Express Router FIFO >1 row in set (0.00 sec) > > >SER start normally if I delete all the records on this table. But I need >create the aliases again, and this is not the normal behaviour..I think. > >Here the description of aliases table: > >mysql> desc aliases; >+---------------+---------------------+------+-----+---------------------+-------+ >| Field | Type | Null | Key | Default > | Extra | >+---------------+---------------------+------+-----+---------------------+-------+ >| username | varchar(64) | NO | PRI | > | | >| domain | varchar(128) | NO | PRI | > | | >| contact | varchar(255) | NO | PRI | > | | >| received | varchar(255) | YES | | NULL > | | >| expires | datetime | NO | | 2020-05-28 21:32:15 >| | >| q | float(10,2) | NO | | 1.00 > | | >| callid | varchar(255) | NO | | Default-Call-ID > | | >| cseq | int(11) | NO | | 42 > | | >| last_modified | timestamp | NO | | CURRENT_TIMESTAMP > | | >| replicate | int(10) unsigned | NO | | 0 > | | >| state | tinyint(1) unsigned | NO | | 0 > | | >| flags | int(11) | NO | | 0 > | | >| user_agent | varchar(50) | NO | | > | | >+---------------+---------------------+------+-----+---------------------+-------+ >13 rows in set (0.01 sec) > > >Important: when "location" table keeps some records, ser doesn't start too: > >mysql> select * from location\G >*************************** 1. row *************************** > username: 579981 > domain: mydomain.com > contact: sip:579981@10.10.60.115:5060 > received: NULL > expires: 2009-02-05 11:46:16 > q: -1.00 > callid: 1f46c06d5e034425286f095b11cf4a1a(a)10.10.60.115 > cseq: 6 >last_modified: 2009-02-05 11:45:46 > replicate: 0 > state: 0 > flags: 0 > user_agent: CM5K-TA2S (807100.02) >1 row in set (0.00 sec) > >Here too I must to delete all the records to let SER start successfully. > >Does anyone knows about this issue? > >Regards, >Claudio > > >------------------------------ > >Message: 3 >Date: Thu, 05 Feb 2009 17:13:09 -0200 >From: caio <elcaio(a)gmail.com> >Subject: Re: [Serusers] preload_udomain error (solved) >To: serusers(a)lists.iptel.org >Message-ID: <498B3A45.5070501(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >caio escribi?: >> Hi all, >> >> Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) >> >> Have this problem when I stop ser service, and then start it again (ser >> -E -D).. >> >> 0(0) preload_udomain: ERROR: bad callid record in table aliases >> 0(0) preload_udomain: ERROR: for username 579981, contact >> sip:579981@mydomain.com >> 0(0) preload_udomain: ERROR: skipping... >> >[cut] > >solved. >there was an old installation of mysql libs (in /usr/sfw/), then I >removed it, and finally recompile mysql.so module including correct >mysql.h (/opt/csw/). > >cheers, >caio > > >------------------------------ > >Message: 4 >Date: Thu, 05 Feb 2009 17:18:03 -0200 >From: caio <elcaio(a)gmail.com> >Subject: Re: [Serusers] Error loading "trusted" table (permissions.so) >To: serusers(a)lists.iptel.org >Message-ID: <498B3B6B.20606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Miklos Tirpak escribi?: > >>> Seem that VAL_TYPE of "val + 2" isn't STRING, read it as DOUBLE >>> (DB_DOUBLE = 1). >>> Do not know why. >>> And because of this the 'if' conditions never are satisfy on trusted.c >>> (line #396): >>> >>> if ((ROW_N(row + i) == 3) && >>> (VAL_TYPE(val) == DB_STRING) && !VAL_NULL(val) && >>> (VAL_TYPE(val + 1) == DB_STRING) && !VAL_NULL(val >>> + 1) && >>> (VAL_TYPE(val + 2) == DB_STRING) && !VAL_NULL(val >>> + 2)) { >> >> >> Thank you very much for the debug lines! I think SER simply uses the >> field types returned by the mysql server, so there might be a problem >> with the server or library. You can check what the field types are along >> with their names returned by the server in modules/mysql/res.c, search >> for the mysql_fetch_fields function. >> >> Miklos >> > >Miklos, > >I've resolved this issue and a similar one ("preload_udomain error"). >There was an old installation of mysql libs (in /usr/sfw/), then I >removed it, and finally recompile mysql.so module including correct >mysql.h (/opt/csw/). > >Now fields returned are with their correct type. > >Thank you.. > >Caio > > >------------------------------ > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > >End of Serusers Digest, Vol 70, Issue 4 >***************************************

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[Kamailio-Users] mediaproxy
by NYam 14 Feb '09

14 Feb '09

Dear Sir, As it was stated in your website, KAMAILIO (OpenSER) is available in binary format for several embedded platforms, including Linksys NSLU. Does it mean that mediaproxy is already installed into the platform with KAMAILIO? If not, how mediaproxy will be installed? Will it be hard to do so? with kind regards Nyam Sharibu

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[Kamailio-Users] SIREMIS feedback
by Elena-Ramona Modroiu 14 Feb '09

14 Feb '09

Hi, it seems that meeting at FOSDEM brought lot of ideas around. Stefan already had a good suggestion - integration of SIREMIS with SEMS giving the possibility to monitor SEMS servers in the same nice manner from a web page. I would like to collect suggestions about what would be good to have in SIREMIS, what should be improved. My goal up to Kamailio 1.5 release is to add support for carrierroute, a cdr rating engine and cleanup the source tree in order to upload the sourceforge svn - this would make easier for others to contribute. However, if it is something we find very useful then I will prioritize. There is time to add new features after the release, as well. Please shoot here any crazy idea you have for a web management interface of Kamailio and later SIP Router. Regards, Ramona

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[Kamailio-Users] UDP packet loss
by Alex R.S.M 13 Feb '09

13 Feb '09

Hi, I am performing SIP to SIP performance testing using Kamailio proxy (UAC-Kamailio-UAC). After increasing the load to 300 calls/sec Kamailio begins dropping UDP packets. (250 cps works fine). CPU utilization and network bandwidth looks fine. I think the bottleneck is the Linux UDP buffer size of the proxy server. Here is the proxy server information: $ uname -a Linux lab-linux01 2.6.24-19-server #1 SMP Wed Jun 18 14:44:47 UTC 2008 x86_64 GNU/Linux $ sudo sysctl -a | grep net.core.rmem_max net.core.rmem_max = 131071 The question is if I increase the linux UDP buffer size do I need to change any configuartion parameter (like MAX_RECV_BUFFER_SIZE) in Kamailio? Thanks, Alex R _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_022…

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[Kamailio-Users] Trying to compile openser 1.3.3 without PKG_MALLOC support
by toly 13 Feb '09

13 Feb '09

Greetings, Trying to compile openser 1.3.3 without PKG_MALLOC, just using OS malloc Here's me Makefile.defs DEFS+= $(extra_defs) \ -DNAME='"$(MAIN_NAME)"' -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' \ -DOS='"$(OS)"' -DCOMPILER='"$(CC_VER)"' -D__CPU_$(ARCH) -D__OS_$(OS) \ -D__SMP_$(ISSMP) -DCFG_DIR='"$(cfg-target)"'\ -DSHM_MEM -DSHM_MMAP \ #-DPKG_MALLOC \ -DUSE_IPV6 \ -DUSE_MCAST \ -DUSE_TCP \ -DDISABLE_NAGLE \ -DHAVE_RESOLV_RES \ -DSTATISTICS \ -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DQM_JOIN_FREE \ #-DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ #-DNO_DEBUG \ #-DNO_LOG \ #-DVQ_MALLOC \ #-DDBG_LOCK \ #-DNOSMP \ #-DEXTRA_DEBUG \ #-DUSE_SHM_MEM It's bailing out compiling blacklists.c blacklists.c: In function `rm_dups': blacklists.c:435: error: syntax error before "else" blacklists.c:440: error: syntax error before "else" blacklists.c: At top level: blacklists.c:443: error: syntax error before "else" blacklists.c:445: warning: type defaults to `int' in declaration of `q' blacklists.c:445: error: invalid type argument of `->' blacklists.c:445: warning: data definition has no type or storage class blacklists.c:446: error: syntax error before '}' token make: *** [blacklists.o] Error 1 Did anybody had a problem compiling openser without pkg_malloc support? I would appreciate any help. TIA Toly. -- View this message in context: http://www.nabble.com/Trying-to-compile-openser-1.3.3-without-PKG_MALLOC-su… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Kernel Message
by Brian Chamberlain 12 Feb '09

12 Feb '09

Hi All, Is this something I should worry about? Feb 12 20:27:22 sw1 kernel: [3273100.087148] INFO: task kamailio:1845 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.087206] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.087291] kamailio D 8ce9265a 0 1845 1829 Feb 12 20:27:22 sw1 kernel: [3273100.087346] f6a98da0 00000082 23f7e426 8ce9265a 000701ae f6a98f2c c302bfa0 00000002 Feb 12 20:27:22 sw1 kernel: [3273100.087427] c01344f6 001e17ce 00000001 00000000 c01190e6 f6a98da0 f6a98da0 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.087538] f74eea84 f74eea8c f74eea88 f6a98da0 c02b8846 f68c3d68 f74eea8c f6a98da0 Feb 12 20:27:22 sw1 kernel: [3273100.087648] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.087730] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.087784] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.087840] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.087896] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.087945] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.087998] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.088055] [<c011845d>] __wake_up_common+0x2e/0x58 Feb 12 20:27:22 sw1 kernel: [3273100.088109] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.088165] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.088223] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.088273] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.088327] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.088387] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.088438] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.088497] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.088554] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.088610] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.091119] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.091174] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.091218] INFO: task kamailio:1846 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.091274] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.091358] kamailio D 00000002 0 1846 1829 Feb 12 20:27:22 sw1 kernel: [3273100.091413] f6a995e0 00000086 00000000 00000002 00000000 f6a9976c c3017fa0 00000000 Feb 12 20:27:22 sw1 kernel: [3273100.091495] ffffffff 1d621a39 00000000 00000000 00000000 00000000 00000000 000000ff Feb 12 20:27:22 sw1 kernel: [3273100.091605] f74eea84 f74eea8c f74eea88 f6a995e0 c02b8846 f74eea8c f6b21d68 f6a995e0 Feb 12 20:27:22 sw1 kernel: [3273100.091715] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.091790] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.091847] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.091895] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.091949] [<c0129ab0>] lock_timer_base+0x19/0x35 Feb 12 20:27:22 sw1 kernel: [3273100.092002] [<c0129b10>] try_to_del_timer_sync+0x44/0x4a Feb 12 20:27:22 sw1 kernel: [3273100.092057] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.092108] [<c01297ec>] process_timeout+0x0/0x5 Feb 12 20:27:22 sw1 kernel: [3273100.092163] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092219] [<c015a0d0>] __rmqueue +0x13/0x174 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.092221] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.092221] INFO: task kamailio:22820 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.092221] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.092221] kamailio D 854aa937 0 22820 22803 Feb 12 20:27:22 sw1 kernel: [3273100.092221] f762fa40 00000086 2603ada4 854aa937 000767a5 f762fbcc c3021fa0 00000001 Feb 12 20:27:22 sw1 kernel: [3273100.092221] c01344f6 001fcdbd 00000001 00000000 c01190e6 f762fa40 f762fa40 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.092221] f6b38544 f6b3854c f6b38548 f762fa40 c02b8846 f68a3d68 f6b3854c f762fa40 Feb 12 20:27:22 sw1 kernel: [3273100.092221] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.092274] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.092328] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092383] [<c015a05d>] __rmqueue_smallest+0x83/0xe3 Feb 12 20:27:22 sw1 kernel: [3273100.092438] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.092493] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092551] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092601] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092653] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.092712] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.092764] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.092822] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.092881] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.092937] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.092991] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.093046] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.093090] INFO: task kamailio:22821 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.093146] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.093230] kamailio D 8543066c 0 22821 22803 Feb 12 20:27:22 sw1 kernel: [3273100.110119] f762ede0 00000082 2603ada4 8543066c 000767a5 f762ef6c c3017fa0 00000000 Feb 12 20:27:22 sw1 kernel: [3273100.110201] c01344f6 001fcdbd 00000002 00000000 c01190e6 f762ede0 f762ede0 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.110312] f6b38544 f6b3854c f6b38548 f762ede0 c02b8846 f6b3854c f6bfdd68 f762ede0 Feb 12 20:27:22 sw1 kernel: [3273100.110422] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.110493] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.110543] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.110598] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.110653] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.110702] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.110755] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.110810] [<c011845d>] __wake_up_common+0x2e/0x58 Feb 12 20:27:22 sw1 kernel: [3273100.110864] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.110919] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.110976] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.111027] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.111079] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.111138] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.111189] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.111248] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.111307] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.111362] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.111417] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.111471] =======================

3 12

[Kamailio-Users] SIREMIS v0.9.1 Released
by Elena-Ramona Modroiu 12 Feb '09

12 Feb '09

Hi everybody, a new version, 0.9.1, of SIREMIS - the web management interface for Kamailio (OpenSER) and SIP-Router.org - is released. New features and improvements: * send MI commands to Kamailio (OpenSER) via UDP * display the output of a MI command * ability to predefine MI commands that can be selected from a drop-down menu * create graphic charts from statistic data * one or more charts per page * one or more graphics per chart * now new coding required, only configuration changes required to add new charts * drop-down selection for user AVP type Goals for next version: * carrierroute management * rating engine for CDRs View screenshots of the 0.9.1: http://www.asipto.com/gallery/v/siremis/ Relevant screenshots for new features: - charts: http://www.asipto.com/gallery/v/siremis/siremis_12.jpg.html - MI: http://www.asipto.com/gallery/v/siremis/siremis_15.jpg.html Demo site (it works on a database with random data, no MI; username: admin, password: admin): http://siremis.asipto.com/demo/ Download: http://siremis.asipto.com/pub/downloads/siremis/ Install/Upgrade documentation: http://siremis.asipto.com/install/ Regards, Ramona

3 3

[Kamailio-Users] how to work with Remote-Party-ID ?
by BERGANZ François 12 Feb '09

12 Feb '09

hello, i need to get the Remote-Party-ID in the header. I have that: Remote-Party-ID: "francois bgz" <sip:170725014@217.64.x.x:5067>;screen=yes;privacy=off;party=calling;-call-i nitiator=5016;-call-initiator-location=int;-redirected-by;-int-ext=5016;-ent -name=Acro;-direction=ext;-call-id=55665. I need to have -ent-name=Acro but, when I use $re, I have sip:170725014@217.64.x.x:5067 how can I do ? thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 2

Re: [Kamailio-Users] [Kamailio-Devel] SIREMIS v0.9.1 Released
by Elena-Ramona Modroiu 12 Feb '09

12 Feb '09

Hi Stefan, Stefan Sayer wrote: > Hi, > > o Elena-Ramona Modroiu [02/11/09 11:48]: > >> Hi everybody, >> >> a new version, 0.9.1, of SIREMIS - the web management interface for >> Kamailio (OpenSER) and SIP-Router.org - is released. >> > this looks very nice. > thanks! > Just recently I started with some monitoring support for SEMS > (http://lists.iptel.org/pipermail/semsdev/2009-February/003517.html) > The way this works is to push some call related info to the monitoring > module, and access it from the outside via DI functions, e.g. via XMLRPC > (http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.monitoring). So you > can get e.g. a list of callids by executing the XMLRPC function > server.listActive(), and get the info for a call with server.get(callid). > > At the moment it logs ruri/from/to/application/direction in the core, > for conference application additionally the conference room. To > illustrate, you would get this from the server: > >>> s.list() > ['3c2d4bde0d60-gds81g00uz1n'] > >>> s.get(s.list()[0]) > [{'from': '"2(a)stefan-lap.office.iptego.de" > <sip:2@stefan-lap.office.iptego.de>', 'app': 'conference', 'to': > '<sip:35@stefan-lap.office.iptego.de;user=phone>', 'conf_id': '35', > 'ruri': 'sip:35@stefan-lap.office.iptego.de;user=phone', 'dir': 'in'}] > > This is of course extensible, e.g. codec used, call start/end time etc. > > Maybe, if we would add a way to monitor SEMS servers from within > SIREMIS, it would give a good opportunity to monitor e.g. conference > bridge, voicemail server, prepaid app, or b2bua (like sbc). > > What do you think? > I think it would be very useful. It looks to me more like an extension to communicate via XMLRPC with SEMS - send command and display the result. What is the XMLRPC you are using in SEMS? The C library used in Kamailio proved quite unstable, therefore SIREMIS uses MI UDP - simpler protocol and easy to troubleshoot. There should be XMLRPC PHP lib, therefore easy to extend SIREMIS. Is it possible to get statistic information out of SEMS? Like number of active calls? Such data can be used to build charts, which is nice to monitor the service health and evolution. Regards, Ramona

2 2

[Serusers] Forking Proxy
by Amanda Mattiuz 12 Feb '09

12 Feb '09

Hello all, Is is possible to configure SER to act as a forking proxy, where the INVITE messages that arrive at the server are forwarded to multiple clients ins parallel? I saw something about call hunt, but it seems that the INVITES are sent sequentially, could I make it become concurrently? Thanks, Amanda

2 1

[Kamailio-Users] IP SEC tunel
by ALAEDDINE abbech 11 Feb '09

11 Feb '09

hi, How can i do a tunel (use IP SEC)whith my serveur openser. is there a code to write to the configuration file to do this. thanks __________________________________________________________________________________________________ Ne pleurez pas si votre Webmail ferme ! Récupérez votre historique sur Yahoo! Mail ! http://fr.docs.yahoo.com/mail/transfert_mails.html

2 1

[Kamailio-Users] Kamilio Failover Options
by Stagg Shelton 11 Feb '09

11 Feb '09

I am looking for some ideas on how to create a more redundant setup in my operation. About 60 days ago our openser 1.3.2 system started sending out 500 errors. Looking through the log file I found alot of ERROR:tm:new_t: out of mem that corresponded to the INVITES. A simple restart of openser freed the memory and calls processed normally as expected. Since then we have upgraded to openser 1.3.3 and haven't experienced the problem since. Currently I am testing kamilio 1.4.3 and planning my migration into kamilio from openser. What I would like to create is a system that has two kamailio systems where one can fail to the other in the event of an application error that prevents calls from processing. Does anyone have any suggestions on implementing a redundant kamilio system, or can point me in the direction of where to start reading and researching. Thanks Stagg Shelton www.3pconnect.com

3 2

Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 11 Feb '09

11 Feb '09

Hello, On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > Hi, > > Just wanted to touch base on the parser to find out if you are still > pursuing this. > it is still on my todo list, however didn't make it in the upcoming 1.5.0. Ovidiu Sas did some commits to the SDP parser and I know couple of others using it with success. Please open a feature request at: http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse It is easier to track and not forgot about it for the next release. Thanks, Daniel > I am in the mind of purchasing some Cisco equipment to do the Sip-MAP > gateway from a GSM network. > I have someone else from the OpenBTS with me on this. > The object is to provide a full fledged sip based routing of voice, sms and > multimedia calls from gsm handsets to BTS with a Sip server backend. > > Cheers, > Abdul Hakeem > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 03 September 2008 17:45 > To: Kristian Kielhofner > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 09/03/08 19:10, Kristian Kielhofner wrote: > >> Hello everyone, >> >> Feel free to set $OSSPROXY to whatever you like = >> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >> >> $OSSPROXY is my favorite SIP toolkit. In addition to being an >> excellent proxy and registrar it serves as a platform to do SIP >> mangling and wrangling. With that in mind let me propose a crazy >> idea... >> >> How hard would it be to get $OSSPROXY to be able to parse what is >> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >> no business getting too involved in the SDP but I feel it makes sense >> in some instances (like this one). What if I want to make a routing >> decision based on an encapsulated ISUP parameter of some sort? Who >> knows what it could be!?!? Maybe there are some other uses but that >> makes the most sense and seems to be the most practical. It should be >> possible to develop an $OSSPROXY module to: >> >> - Use the SDP parser to find the ISUP SDP part: >> >> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >> >> - Pass that part into a "real" ISUP parser (if necessary). I couldn't >> seem to find the perfect library but perhaps something like this could >> work: >> >> http://www.openss7.org/isup.html >> >> - Make the ISUP parameters available via AVPs, pseudo variables, etc >> for use in the $OSSPROXY script. >> >> I don't really have a need for this but it seems like it could be >> cool. Am I nuts? >> >> >> > I would expose the body as a dynamic structure, so components will be > accessible in the script, e.g. (naming schema just for example), > > if($body(0=>content-type)=="application/sdp") > { > if($body(0=>m=>type) == "audio") > { > } > } > > Based on the available content type parsers (sdp, isup), the structure has > different attributes. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Looking for a highly available / load balanced setup
by John D 11 Feb '09

11 Feb '09

I currently have three Asterisk servers in my environment. None of them speak to each other just yet. The idea is to install Kamailio and be able to load balance/distribute SIP calls to any of the three asterisk boxes. If an entire box plug gets pulled, the operation continues. If a Kamailio box gets the plug pulled (and due to limited resources, I plan at first to run kamailio on all three of the asterisk boxes), the operation will continue. Does anyone have good documentation or insight in order to facilitate this sort of setup? Thanks.

4 6

[Kamailio-Users] New developer: Don Jackson
by Henning Westerholt 10 Feb '09

10 Feb '09

Hi all, its my pleasure to announce a new developer: Don Jackson. He'll maintain in the future the OpenBSD port of our favorite SIP server, and make sure that Kamailio will work there as good as on other operating systems. His sf.org id is 'dcjackson'. Thank you Don for the packaging you've done, i'm looking forward to work further with you. Best regards, Henning

1 0

[Kamailio-Users] OpenBSD port for Kamailio 1.4.3
by Don Jackson 10 Feb '09

10 Feb '09

Hello, I have updated my OpenBSD port to support Kamailio 1.4.3 I sent the updated port to the ports@openbsd mailing list: http://archives.neohapsis.com/archives/openbsd/2009-01/1107.html In addition, the port is available here: http://www.kamailio.org/pub/kamailio/1.4.3/packages/openbsd Thanks to Henning Westerholt for putting the files up on kamailio.org

1 0

Re: [Kamailio-Users] [Kamailio-Devel] FOSDEM2009 - first remarks/photos
by Daniel-Constantin Mierla 10 Feb '09

10 Feb '09

On 02/10/2009 04:44 PM, Iñaki Baz Castillo wrote: > 2009/2/10 Johansson Olle E <oej(a)edvina.net>: > >>> I have been to XMPP interop, although they focused on Voice/Jingle, I >>> tried the new purple module -- still some things to sort out and I will >>> post a resume about this particular case and entire event. >>> >> I spent monday on the XMPP dev meeting a got a few ideas for improvements. >> > > Nooo, forget XMPP!! > SIP SIMPLE already has enough cool specifications, the problem is that > nobody implements them! > :-) -- maybe SIP extension writers can learn from xmpp where they go first for prototype implementation and then to standardization. You are right, there are cool specification, but many do not map properly to reality. > Looking for interoperability with XMPP will not help, and I don't want > a future in which my softphone implements SIP for voice and XMPP for > presence/IM, that's a dirty workaround! > I think a competing open protocol is good all the time, makes things improve in both sides. Apart of that, sip and xmpp does not really match, SIP is more a routing-like protocol, while xmpp is a content-like protocol. XMPP network exist and cannot be ignored. A good usability of xmpp protocol will be communication between various points of your environment, sip servers, web servers, etc... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] FOSDEM2009 - first remarks/photos
by Daniel-Constantin Mierla 10 Feb '09

10 Feb '09

Hello, thanks to Alexandr Dubovikov, a selection of photos from FOSDEM2009 and Kamailio/SIP Router Meeting are now available at: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ Some movies should become available as well. As for the FOSDEM itself, there were a lot of sessions focusing on scalability for backends, mainly from MySQL track. There are some goodies that will help improving performance with 0-lines of code in the sip servers. I have been to XMPP interop, although they focused on Voice/Jingle, I tried the new purple module -- still some things to sort out and I will post a resume about this particular case and entire event. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] dialog enhancements #3
by Daniel-Constantin Mierla 09 Feb '09

09 Feb '09

Hello, the dialog module for upcoming release kamailio/openser 1.5.0 (http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x) has the ability to execute a script route when a dialog timeouts. Just set the $dlg_ctx(timeout_route) before creating the dialog. route { ... $dlg_ctx(timeout_route) = 11; dlg_manage(); ... } route[11] { ... } Before the execution of timeout route the $dlg(...) psudo-variable is set to the expired dialog. One particular use case is to write full cdr, via sqlops and using $dlg(...) pseudo-variable. All functions that can be executed in a request route are safe here - there is a faked SIP message given as parameter. There are couple of other new properties that can be set before dialog creation: - send BYE at timeout - to make that happen you have to set: $dlg_ctx(timeout_bye) = 1; - set an integer that can be used as a bitmask flag checking later: $dlg_ctx(flags) = ...; Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] errors installing on centos 5 64 bits
by Fabian Borot 06 Feb '09

06 Feb '09

Hello I am trying to install it on Centos 5 64 bits using this command: NICER=1 make prefix=/usr/local/kamailio-install-dir all and it fails with this message: make[1]: Entering directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' Linking db_mysql.so /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make[1]: *** [db_mysql.so] Error 1 make[1]: Leaving directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' make: *** [modules] Error 2 I think that somehow it is trying to use the lib for 32 bits or that my specific lib file is not what it should be or it is not on my system. I have the folders /usr/lib and usr/lib64 with its respective library files. Is this "libssl" related to openssl? I have these installed on my system already: [root@proxy1 lib64]# yum install openssl Setting up Install Process Parsing package install arguments Package openssl - 0.9.8b-10.el5_2.1.x86_64 is already installed. Package openssl - 0.9.8b-10.el5_2.1.i686 is already installed. Nothing to do [root@proxy1 lib64]# I also looked for "libssl" and this is what I see, it was only found in the folder for 32 bits libs if I am not mistaken. if that is the case how can i get the 64 bits equivalent? [root@proxy1 lib64]# find / -name "libssl.so" /usr/lib/libssl.so [root@proxy1 lib64]# thank you very much Fabian _________________________________________________________________ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_AE_Faster_022009

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Re: [Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas
by Rash 06 Feb '09

06 Feb '09

Hi Daniel, thanks for the quick repsonse, you delivered the magic hint! Putting the count paramaeter into double quotes works. Thanks for your effort! Rash

2 1

[Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas
by Rash 06 Feb '09

06 Feb '09

Hello, I'm using latest version of Kamailio (OpenSER) from daily snapshot, as i wanted to test the rtpproxy_stream2uac function of the nathelper module in order to make announcements with rtpproxy. Integration of Kamailio and RTPProxy was successfull and I added the following line to the kamailio.cfg: rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", -1); as described in the nathelper documentation ( http://www.kamailio.org/docs/modules/devel/nathelper.html#rtpproxy_stream2x… ) After adding this line to the appropriate section (onreply_route in my case) kamctl refuses to start.: kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 29-30: syntax error kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 30-31: bad arguments where line 414 corresponds to the line shown above. I can't find the error as the line is copied from nathelper module. Playing around with the parameters, didn't lead to anything. It would be nice if anyone could help or tryout the example from nathelper documentation and feedback. Best regards Rash

2 1

[Kamailio-Users] dialog enhancements #2
by Daniel-Constantin Mierla 06 Feb '09

06 Feb '09

Hello, another addition to the dialog module is the new pseudo-variable class $dlg(...) http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:devel#dlg_attr With it you can get access to all attributes of current processed dialog, e.g., caller and callee URI, start time, flags... A typical use case is to write full cdr when the BYE comes over - the dialog stores the start time and you can get the end time via $Ts. You can either write the acc record at dialog ending or update the start acc event written when INVITE was processed - you have access to the call-id, from tag and to tag in the right order and one shot of sql_query() will do it. Cheers, Daniel

3 2

[Kamailio-Users] location table
by joy yue 05 Feb '09

05 Feb '09

Hi, Is there a limitation on the number of rows in location table? In my rig, whatever number of users I use (>2million users), I notice the number of users in location table is 343707, which is far less than the number of users in my test. I thought previously that location table saves all the users in my test. Also when I use large number of users (>2million), openser pops up an error: ERROR: registrar:update_contacts: invalid cseq for aor <xxxx>. Has anyone saw this before? I am using openser1.3.2. Thanks, -Joy

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[Serusers] Error loading "trusted" table (permissions.so)
by caio 05 Feb '09

05 Feb '09

Hello all, Does anyone knows about this "permission.so" module error when try to load "trusted" table (mysql)? If "trusted" table is empty, there is no error, but whan at least there is 1 record, the error appears. for example: mysql> select * from trusted; +---------------+-------+--------------+ | src_ip | proto | from_pattern | +---------------+-------+--------------+ | 10.124.35.253 | any | ^sip:.*$ | +---------------+-------+--------------+ and when I try lo restart ser: Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 966737 daemon.info] permissions - initializing Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 414293 daemon.warning] Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 844440 daemon.warning] Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 650101 daemon.crit] init_trusted(): Reload of trusted table failed Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 345573 daemon.error] Error while initializing allow_trusted function Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 660978 daemon.error] init_mod(): Error while initializing module permissions or when executing serctl fifo cmd: root@delcolimba /usr/local # sbin/serctl fifo trusted_dump 200 OK root@delcolimba /usr/local # sbin/serctl fifo trusted_reload 400 Trusted table reload failed root@delcolimba /usr/local # tail /var/adm/messages Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 450600 daemon.error] ERROR: fifo_server: command (trusted_reload) processing failed Thank you, Caio

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[Serusers] preload_udomain error
by caio 05 Feb '09

05 Feb '09

Hi all, Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) Have this problem when I stop ser service, and then start it again (ser -E -D).. 0(0) preload_udomain: ERROR: bad callid record in table aliases 0(0) preload_udomain: ERROR: for username 579981, contact sip:579981@mydomain.com 0(0) preload_udomain: ERROR: skipping... And finally it dies with: 0(0) get_connection(): Connection not found in the pool Fallo de segmentaciÃ³n (core dumped) It happens with many records on aliases table and happens too with only one record: mysql> select * from aliases\G *************************** 1. row *************************** username: 579981 domain: 10.10.1.1 contact: sip:579981@mydomain.com received: NULL expires: 2009-02-05 11:45:21 q: 1.00 callid: The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything cseq: 42 last_modified: 2009-02-05 11:45:46 replicate: 0 state: 0 flags: 128 user_agent: SIP Express Router FIFO 1 row in set (0.00 sec) SER start normally if I delete all the records on this table. But I need create the aliases again, and this is not the normal behaviour..I think. Here the description of aliases table: mysql> desc aliases; +---------------+---------------------+------+-----+---------------------+-------+ | Field | Type | Null | Key | Default | Extra | +---------------+---------------------+------+-----+---------------------+-------+ | username | varchar(64) | NO | PRI | | | | domain | varchar(128) | NO | PRI | | | | contact | varchar(255) | NO | PRI | | | | received | varchar(255) | YES | | NULL | | | expires | datetime | NO | | 2020-05-28 21:32:15 | | | q | float(10,2) | NO | | 1.00 | | | callid | varchar(255) | NO | | Default-Call-ID | | | cseq | int(11) | NO | | 42 | | | last_modified | timestamp | NO | | CURRENT_TIMESTAMP | | | replicate | int(10) unsigned | NO | | 0 | | | state | tinyint(1) unsigned | NO | | 0 | | | flags | int(11) | NO | | 0 | | | user_agent | varchar(50) | NO | | | | +---------------+---------------------+------+-----+---------------------+-------+ 13 rows in set (0.01 sec) Important: when "location" table keeps some records, ser doesn't start too: mysql> select * from location\G *************************** 1. row *************************** username: 579981 domain: mydomain.com contact: sip:579981@10.10.60.115:5060 received: NULL expires: 2009-02-05 11:46:16 q: -1.00 callid: 1f46c06d5e034425286f095b11cf4a1a(a)10.10.60.115 cseq: 6 last_modified: 2009-02-05 11:45:46 replicate: 0 state: 0 flags: 0 user_agent: CM5K-TA2S (807100.02) 1 row in set (0.00 sec) Here too I must to delete all the records to let SER start successfully. Does anyone knows about this issue? Regards, Claudio

1 1

[Serusers] configure SER and SEMS to work together
by Martin Lane 05 Feb '09

05 Feb '09

I am following the 'mini-Howto on how to configure SER and SEMS to work together' guide on iptel.org. So I have a separate directory for integrating the SER and SEMS functionality. For development purposes is there a switch to change the path were SER looks for the modules? As well control which version of SEMS is being used? I've installed Sip Express Media Server (1.1.0-rc1 (i386/linux)) as well as the version from the howto for testing. My basic problem is I currently get a version mismatch error: 0(9590) ERROR: module version mismatch for /usr/local/lib/ser/modules/sl.so; core: ser 0.9.6-sems (i386/linux); module: ser 0.9.6 (i386/linux) The appropriate modules are located in under my ser-sems test dir and I have not been successful in redirecting SER to look there. Also, I have modified ser.cfg and sems.cfg for my system but I did not find a default ser_sems.cfg. Is this file necessary and where is the default location? Thanks for any help or advice. mll

2 1

[Kamailio-Users] dialog enhancements #1
by Daniel-Constantin Mierla 04 Feb '09

04 Feb '09

Hello, during the week before freezing the code for release kamailio (openser) 1.5.0, I have added several enhancements to the dialog module. I am going to present them in several emails, as I get the documentation properly updated. There is now a new function called dlg_bridge(From, To, outbound_proxy). It can be used to connect From and To users. The mechanism is a simple call transfer scenario. kamailio => INVITE (hold) =>From kamailio <= 200ok <= From kamailio => REFER (To) =>From kamailio <= 200ok <= From ... From => INVITE ... =>To ... kamailio => BYE =>From kamailio <= 200ok <= From The function can be called from configuration file or via MI interface. It opens the door for several cute features you can do with the sip server. Here are couple of usage examples: - easy integration with web/external applications for Click-To-Dial. Doing it with kamctl will look loke: kamctl fifo dlg_bridge sip:from@kamailio.org sip:to@kamailio.org - wake up/reminder/scheduled call - use rtimer module to run scheduled calls - you can connect your user to a media server playing a message, to a conference room or where the user sets its preference - call again - when a user is calling a busy destination, when that destination becomes available, the server can initiate the call to connect the caller with the destination Documentation is available at: http://kamailio.org/docs/modules/1.5.x/dialog.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 4

[Kamailio-Users] Reg. SIP conferencing support for RFC 4579
by Padmaja R 04 Feb '09

04 Feb '09

Hi all, Is it possible to use Openser as the SIP conference server? I understand that Openser cannot do the media mixing required of a centralized media server but if it can do the centralized signaling by acting as a focus and let a media proxy do the media mixing, that would be good. I need ad hoc conferencing capability. Is there any media proxy that can achieve this together with openser? My scenario is as follows:- Say A and B are in a SIP call, and A and B suddenly decide to have a conference with C. A puts B on hold and dials C' s number. After call between A and C is up, if A presses hook flash, this sends an invite to the conference application namely: conf@openseratlocalhost:5060 (which is the ip and port number at which my openser proxy would be listening). The server should then send back 200 Ok with the contact header containing the isfocus" parameter of the form -Contact: <sip:conference1@mediaproxyip;isfocus> The isfocus is an RFC 4579 based feature. The 200 Ok responses to other invites (normal calls that do not create conference) should not include this isfocus in the contact header of 200 Ok. Finally, the openser proxy should be able to contact the media proxy on receiving an invite to create the conference and send that ip and focus information in the 200 Ok. The UA -A would then send a REFER to B and C to refer to the media server and they disconnect with A. Then all three UAs would individually send invite to the uri returned in the contact header and join the conference. Thanks, Padmaja

4 4

[Serusers] SER and media server
by Chatrapathi. H. Rahul 04 Feb '09

04 Feb '09

Hi I am using solaris 9v. I tried to use SEMS as media server with SER. SEMS is not compatible with Solaris 9v. Gives compilation errors. My query is whether any other opensource Media Server is available that uses SER and compatible with solaris 9v.This would help me a lot. Thanks in advance. -- Regards Rahul

1 0

[Kamailio-Users] FOSDEM Meeting - Feb 7, 2009, 19:30, Le Mess
by Daniel-Constantin Mierla 04 Feb '09

04 Feb '09

Hello, the place for the social networking event at FOSDEM 2009 has been selected: Le Mess http://www2.resto.be/mess/home-en.htm Louis Schmidtlaan - Av. Louis Schmidt 1 1040 BRUXELLES (ETTERBEEK) It is in walking distance from FOSDEM site. More details at: http://www.kamailio.org/mos/view/News/NewsItem/FOSDEM-2009---Social-Network… If you haven't confirmed your participation, you can still join, subject of available free places in the restaurant. See you Saturday! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] New module : PURPLE - IM/Presence multi-protocol gateway
by Eric PTAK 04 Feb '09

04 Feb '09

Hello everyone, Let me introduce to you a new module, available in trunk since yesterday, so it will be included in 1.5.0 release : >From http://kamailio.org/docs/modules/devel/purple.html: *Purple module is a multi-protocol instant-messaging gateway module which use libpurple, the library on bottom of Pidgin (ex-gaim).* *This plugin was wrote in order to proof the concept of gateway between SIP and MSN/Live networks.* *[...]* *It's no really adapted to a production environment with a huge user count but can be used in order to demonstrate the concept or to use it in SOHO environment.* I hope you'll enjoy purple module, and I'll welcome all feedbacks. Eric.

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Re: [Kamailio-Users] sip trace & duplicate_uri
by Daniel-Constantin Mierla 03 Feb '09

03 Feb '09

On 02/02/2009 09:35 PM, kokoska rokoska wrote: > > [...] >>> Hello, >>> >>> i have increased debug to 6 and I can not see errors in syslog. >>> Messages are traced, stored into DB, but not duplicated. >>> I'm at 1.4.3 on CentOS 5 - if it makes sense... >>> >>> Thank you very much, Daniel, for your interest! >>> >>> >>> >> ok, checked the source code. For some reason I do not recall now, when I >> wrote the module the duplication is done only when traced_user_avp is set. >> http://kamailio.org/docs/modules/devel/siptrace.html#id2452997 >> >> Set that avp and try again. If works I will try to fix and make it work >> without avp. >> >> Thanks, >> Daniel >> >> >> > > > Thank you very much, Daniel, for your help! > > It works :-) > > BTW: My idea is to offload proxy from (blocking ?) DB operatinos and > still have all possible logs => I think I can duplicate all messages to > another kamailio server and store them there to another DB. Is it > doable? With sip trace you still get them in a DB, because it first stores to DB and then sends to another address. However, what you want to achieve is something good, so for the future I will have in mind the option just to replicate to another address and skip writing to db. > Or, better to say, can I trace and duplicate messages on one > server and not log them? > If you want to duplicate the messages that come to kamailio, you can use the send() function in route and onreply_route block. http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:devel#send_destinat… If you want the siptrace module to do it, you have to hack it a bit. Practically is to introduce a new parameter, e.g., skip_db_storage, and put the code that writes in db inside a if not skip_db_storage. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] [Kamailio-Devel] New module : PURPLE - IM/Presence multi-protocol gateway
by Eric PTAK 03 Feb '09

03 Feb '09

I agree with you, the module needs improvements on that point. It's a really old point I missed when we decided to reuse and to reverse that module. I think about two manners in order to solve it : * implement a return code for purple_send_message * sending a reply directly in the module Regarding the design of the module, I think both possibilities needs a lot of mod. I'm not sure how to consider that mod as it's not a bug but a limitation. Is it a new feature I'll code when the trunk will be unfreezed ? If yes, we may add that limitation in the doc for now. Eric. 2009/1/29 Iñaki Baz Castillo ibc(a)aliax.net > Why is needed to reply "200"? doesn't this module act as a proxy > waiting for a response from downstread (i.e. MSN network) and mapping > that response to the appropiate SIP code? > > This is: if I send a MESSAGE to a MSN account not online (or not > existing) I will receive "200" anyway, shouldn't I receive a 480 or > 404? > > Regards. > > > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > > _______________________________________________ > Kamailio (OpenSER) - Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel > http://lists.openser-project.org/cgi-bin/mailman/listinfo/devel

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[Kamailio-Users] wiki documentation index for kamailio
by Henning Westerholt 03 Feb '09

03 Feb '09

Hello all, thanks to Frederick Bullik we've now a nice documentation index for the Kamailio 1.4.x and 1.5.x releases available. You can easily lookup all module exports like functions [1] or module parameters [2]. The index is sorted alphabetically and provides convinient links to jump directly to the respective place in the module documentation. You'll find the new index in our wiki at: http://kamailio.org/dokuwiki/, as well as the old ones for past OpenSER releases. Best regards, Henning Westerholt [1] http://www.kamailio.org/dokuwiki/doku.php/modules:1.5.x:index-functions [2] http://www.kamailio.org/dokuwiki/doku.php/modules:1.4.x:index-parameters -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany I'm going to FOSDEM

2 1

[Kamailio-Users] [Fwd: Re: sip trace & duplicate_uri]
by kokoska rokoska 02 Feb '09

02 Feb '09

Daniel-Constantin Mierla napsal(a): > Hello, > > On 02/02/2009 08:08 PM, kokoska rokoska wrote: >> >> Daniel-Constantin Mierla napsal(a): >> >>> Hello, >>> >>> do you get any error in the syslog? >>> >>> Cheers, >>> Daniel >>> >>> >> >> >> Hello, >> >> i have increased debug to 6 and I can not see errors in syslog. >> Messages are traced, stored into DB, but not duplicated. >> I'm at 1.4.3 on CentOS 5 - if it makes sense... >> >> Thank you very much, Daniel, for your interest! >> >> > > ok, checked the source code. For some reason I do not recall now, when I > wrote the module the duplication is done only when traced_user_avp is set. > http://kamailio.org/docs/modules/devel/siptrace.html#id2452997 > > Set that avp and try again. If works I will try to fix and make it work > without avp. > > Thanks, > Daniel > > Thank you very much, Daniel, for your help! It works :-) BTW: My idea is to offload proxy from (blocking ?) DB operatinos and still have all possible logs => I think I can duplicate all messages to another kamailio server and store them there to another DB. Is it doable? Or, better to say, can I trace and duplicate messages on one server and not log them? Best regards, kokoska.rokoska

1 0

[Kamailio-Users] sip trace & duplicate_uri
by kokoska rokoska 02 Feb '09

02 Feb '09

Hi all, could someone be so kind and explain me how to use (or how exactly works) duplicate_uri in siptrace module? I think, if I setup modparam("siptrace","duplicate_uri","sip:1.2.3.4:5060") that all (and what is it "all"? :-) traced messages will be copied to defined sip uri (1.2.3.4:5060). I see nothing "on the wire" with ngrep, but all messages are succesufly inserted into local DB so I thing the siptrace module works. But doesn't copy messages to 1.2.3.4:5060 :-) Any hitns are really welcome! Best regards, kokoska.rokoska

2 3

[Kamailio-Users] store a double in avp
by ALAEDDINE abbech 02 Feb '09

02 Feb '09

hi all, can i store a double in avp? i try to store a double from my DB in the avp, so i have alwyse null. who have a solution for this pb. thanks

2 2

[Kamailio-Users] FOSDEM 2009 - Kamailio (OpenSER)/SIP-Router Social Networking Event
by Daniel-Constantin Mierla 02 Feb '09

02 Feb '09

Hello, a bunch of us will be at FOSDEM 2009: http://fosdem.org/2009/ Therefore is the opportunity for first Social Networking Event in 2009. Everyone interested in Kamailio (OpenSER), SIP Router, SER, SIP and not only are invited to join -- participation is free, everybody pays for its expenses. The event is very likely to be a dinner on Saturday evening, Feb 7, 2009. So far, from Kamailio (OpenSER) community you can meet there: - Henning Westerholt - http://www.kamailio.org/mos/view/Management/ - Elena-Ramona Modroiu - http://www.kamailio.org/mos/view/Management/ - Daniel-Constantin Mierla - http://www.kamailio.org/mos/view/Management/ - Dan Bogos - http://itsyscom.com/ In order to know properly the size of the place we need, please drop me an email if you intend to attend. People knowing Brussels are welcome to suggest a nice place to meet (good beer and food are a must). Looking forward to meeting you in Brussels. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

[Kamailio-Users] No UDP Address already in use?
by mayamatakeshi 31 Jan '09

31 Jan '09

Hello, If I start a second instance of kamailio by mistake, if the cfg file instructs it to listen to tcp ports, the second instance will exit with something like this: tcp_init: bind(7, 0x81b28ec, 16) on 127.0.0.1:5060 : Address already in use But if the cfg file only instructs kamailio to bind to udp ports, the second instance will not exit. Is this a bug? I'm curious because if we start kamailio and a udp port is in use by another application (not kamailio), kamailio will exit with udp_init: bind(4, 0x819f274, 16) on 192.168.88.29: Address already in use Why is that so? regards, mayama

3 4

[Serusers] Modified "To" header on REGISTER
by caio 30 Jan '09

30 Jan '09

Hello, Does save() method of registrar module use original sip message or can it use a modified one like for example by the subst() method of textops module? For example: A coming "To" header of the a REGISTER is modified and then the save() is called: Code: # REGISTER subst_uri('/^(.*)sip:(ip_address)(.*)$/\1sip:mydomain\3/i'); subst('/^To:(.*)sip:(.*)@(ip_address)(.*)$/To:\1sip:\2\@mydomain\4/i'); save("location"); Log: Jan 12 10:09:34 [/usr/sbin/ser] subst_run: matched (0, 17): [sip:200.2.127.187]_ Jan 12 10:09:34 [/usr/sbin/ser] textops: subst_uri_f: match - old uri= [sip:ip_address], new uri= [sip:mydomain]_ Jan 12 10:09:34 [/usr/sbin/ser] subst_run: running. r=0_ Jan 12 10:09:34 [/usr/sbin/ser] subst_run: matched (172, 40): [To: "579997" <sip:579997@ip_address>_]_ Jan 12 10:09:34 [/usr/sbin/ser] textops: subst_f: replacing at offset 208 [To: "579997" <sip:579997@ip_address>_] with [To: "579997" <sip:579997@mydomain>_]_ Cheers.. Caio

1 1

[Kamailio-Users] STUN and TM
by Jason Penton 30 Jan '09

30 Jan '09

Hi *, What is the logic behind the transation matching in Kamailio. I am finding a problem whereby a call setup transaction (INVITE, 180 Ringing, PRACK, 200OK, ACK). between 2 Snom phones works in normal mode. However, when one phone is behind a NAT with STUN support, everything works perfectly (i.e. call setup, 2-way audio, etc), BUT the tm module doesnt match the transaction between the original invite and the subsequent messages (180, 200ok, etc). the impact here is that the fr_inv_timer eventually fires even though the call has been answered. Note: this only happens when the call is destined for the device behind the NAT. Any ideas. Perhaps an explanation on how the tm module does transaction manager. RFC3261 states that transaction matching is done based on the via branch parameter. Thanks in advance Jason

2 7

[Kamailio-Users] code frozen for 1.5.0
by Daniel-Constantin Mierla 29 Jan '09

29 Jan '09

Hello, starting with today, the release cycle for Kamailio (OpenSER) v1.5.0 enters in testing phase. Like in the past, this mean no new features will be added for a while, the focus is on testing, bug fixes and documentation improvements. I can say, being here from day one, this upcoming release had a shorter development period, but it is the one with the largest new features set. See a summary here: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x A special care was given to code cleanup and core. The core got slimmer - lot of config-related features were moved to modules, therefore now it is less exposed to bugs and more performant. From the project's environment: - we got three new developers - a brand new web management interface - http://siremis.asipto.com - next week will be a meeting in Brussels, at Fosdem with many developers showing up there: http://www.kamailio.org/mos/view/News/NewsItem/FOSDEM-2009---Social-Network… Worth to mention that this is the last release based on current core, next ones will use the SIP-Router.org core framework: http://sip-router.org, that will make possible to combine all the features from Kamailio (OpenSER), SIP Express Router (SER) and OpenIMSCore projects in a single application, in a collaboration environment sustained by the developers from all three projects. First I want to thank to all developers for the work and time dedicated to develop version 1.5.0 and to our community members for valuable feedback and testing. That being said, I invite everyone to test and report any issue found. To get the source code from SVN use: svn co https://openser.svn.sourceforge.net/svnroot/openser/trunk kamailio Please report to devel(a)lists.kamailio.org or project's tracker: http://sourceforge.net/tracker/?group_id=139143 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] textops: new function - set_rpl_body(...)
by Daniel-Constantin Mierla 29 Jan '09

29 Jan '09

Hello, few days ago I added a new function to textops module: - set_rpl_body(body, content-type) It can be used to set the body of a reply generated by kamailio/openser. route { if(is_method("HELLO") && $ru == "sip:server@kamailio.org") { set_rpl_body("Hi $fn!", "text/plain"); send_reply("200", "ok"); exit; } The insert_body() function (also new in 1.5.0) has been renamed to set_body() for better coherence. This one can be used to set the body of current processed sip message. An an example, set_rpl_body() can be used for pull-like data transfer between servers. Using uac module (and eventually rtimer module), servers in a farm can interrogate each other and exchange data. Some useful links: http://kamailio.org/docs/modules/1.5.x/uac.html#id2467913 http://kamailio.org/docs/modules/1.5.x/rtimer.html http://kamailio.org/docs/modules/1.5.x/textops.html#id2468015 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] SER Consultants in US
by Tichner, Jeromy E (US SSA) 29 Jan '09

29 Jan '09

Hello, We're looking for some phone support from a consulting firm for a SER configuration. Looking for an hourly estimate and must be available ASAP. Must be a US citizen. Regards, Jeromy

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[Kamailio-Users] ldap in kamailio
by Husna Bte Mad Baguri 29 Jan '09

29 Jan '09

Help, How to install ldap module in kamailio Regards Husna ________________________________ This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

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[Kamailio-Users] detecting the presence of a header
by Daniel Corbe 28 Jan '09

28 Jan '09

I need to quickly detect the presence of the Alert-Info header and do special processing if it is present. Is there facilities to do this currently? -Daniel

2 1

[Kamailio-Users] function to send ACK message
by Alex R.S.M 28 Jan '09

28 Jan '09

Hi, I need kamailio proxy to reply with an ACK to 200 OK, i.e. the proxy stays in the call: UAC sends INVITE to kamailio, kamailio forward it to UAS UAS sends 180 to kamailio, kamailio forward it to UAC UAS sends 200 to kamailio, kamailio forward it to UAC AND sends ACK to UAS I was wondering what function I can use so kamailio can generate an ACK message to response to 200 OK. Apparently both t_reply () and sl_send_reply() functions create responses and not an ACK message. Thanks, Alex _________________________________________________________________ Windows Live™ Hotmail®…more than just e-mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t2_hm_justgotbetter_how…

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[Kamailio-Users] new user
by Andy Nedelcu 28 Jan '09

28 Jan '09

Hey, I am new at working with SIP Servers and I was wondering if Kamailio has the same features as Asterisk. I have bought a PCI card for connecting to the phone line and I want to know how can I make Kamailio aware of this card or a simple phone modem. I use FreeBSD and the only version that I found was OpenSer 1.2. Because I love being me and I love loving her, SC BMS SRL - Departamentul IT - Nedelcu Andrei

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[Kamailio-Users] Database error: failed to request mysql lock ()
by Brian Del Shasta 27 Jan '09

27 Jan '09

Database error: failed to request mysql lock () Has anyone else experienced this issue?

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[Kamailio-Users] Silently replying 500 ?
by Aurelien Grimaud 27 Jan '09

27 Jan '09

Hi, I have a route block which issues a t_relay(). route[1] { t_relay() ; } If this t_relay fails, is there a default reply from openser to caller ? Let me explain. In my logs, I have an incoming INVITE, followed 2 ms after t_relay by an ACK from uac. The callee never gets the INVITE. UAC -- INVITE --> openser : t_relay() UAC -- ACK --> openser I guess UAC gets a default reply from openser, I want to make sure of it. I have no ERROR in my log file. UAC logs tells that openser hang up with a server internal error (no sip trace, no sip code, just an indication of what cause could be). This is openser 1.2.0 Is anyone aware f this issue ? Could someone point me where the 5xx reply is in source code ? If it is a default reply, is there now an ERROR message before reply ? Thanks, Aurelien

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[Kamailio-Users] openSER concept question
by Alex R.S.M 27 Jan '09

27 Jan '09

Hi, In a SIP environment with SIP end points (UAC, UAS) and several SIP proxy hubs including openSER, is each hub like openSER supposed to reply with an ACK to 200 OK and BYE messages coming from end points? (i.e. openSER should stay on the call) Or each hub supposed to only propagate all messages including ACKs ? Thanks, Alex _________________________________________________________________ Windows Live™ Hotmail®…more than just e-mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t2_hm_justgotbetter_how…

2 1

[Kamailio-Users] error, please help me
by Daniel Kemayou 26 Jan '09

26 Jan '09

Hello all I have a error when i finish the setup of SIREMIS, i have window with > userid and password when i enter the userid and password i have this > error > > [2009-01-24 19:11:48 (GMT)] An exception occurred while executing this > script: > Error message: #0, The mysql driver is not currently installed Script > name and line number of error: > /var/www/siremis-0.9.0/openbiz/others/zendfrwk/library/Zend/Db/Adapter > /Pdo/Abstract.php:104 > > can you help me please DISCLAIMER: This email and any attachments are confidential and are intended solely for the addressee. If you are not the addressee tell the sender immediately and destroy it. Do not open, read, copy, disclose, use or store it in any way, or permit others to do so. Emails are not secure and may suffer errors, viruses, delay, interception, and amendment. Creolink Communications and its subsidiaries do not accept liability for damage caused by this email and may monitor email traffic. DENI : Ce mail, y compris toute piece jointe, est confidentiel. Il est exclusivement adresse aux destinataires designes. Si vous le recevez par erreur, veuillez immediatement aviser l'expediteur et detruisez-le sans l'ouvrir ni le lire. Toute copie, diffusion, conservation, utilisation sous quelque forme que ce soit ou l'autorisation de toute personne a le faire est interdite. Le mail n'est pas un moyen sur et peut contenir des erreurs ou des virus, comme il peut etre retarde, intercepte ou modifie. Creolink Communications et ses filiales declinent toute responsabilite pour des dommages causes par ce mail et se reservent le droit de controler la circulation des mails.

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[Kamailio-Users] carrierroute
by michel freiha 26 Jan '09

26 Jan '09

Hi all, I have a problem with carrierroute module...I have configured it as follow in my config file: loadmodule "carrierroute.so" ######Carrier Route modparam("carrierroute", "db_url", "mysql://opensips:123456@MYSQL_IP /opensips") modparam("carrierroute", "config_source", "db") In the route function i have created the following function: if (!cr_route("default", "0", "$rU", "$rU", "call_id")) { sl_send_reply("403", "Not allowed"); } else { # In cas of failure, re-route the request t_on_failure("1"); # Relay the request to the gateway t_relay(); } When trying to restart kamailio server I got a the below error: Jan 24 11:10:22 [12841] DBG:core:fix_actions: fixing sl_reply_error, line 381 Jan 24 11:10:22 [12841] DBG:core:fix_actions: fixing cr_route, line 390 Jan 24 11:10:22 [12841] ERROR:core:pv_parse_spec: bad parameters Jan 24 11:10:22 [12841] INFO:carrierroute:carrier_fixup: carrier tree default has id 3 Regards

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[Kamailio-Users] DNS unreachable
by Zappasodi Daniele 26 Jan '09

26 Jan '09

Hello, I'm testing the proxy behaviour (Openser 1.3.2) when DNS servers are unreachable and I observe that with few requests that require the DNS, I have the proxy completely blocked, while it could serve all the other methods that don't need the DNS (method with IP adddress or incoming REGISTERs). I observe that each child process remains suspended until all DNS requests are finished (NAPTR, SRV and A). If I configure two DNS server and X retransmissions after Y seconds, the child remain suspended for: 2 * X * Y * 3 seconds (3=NAPTR+SRV+A queries) Even if I set low values for X and Y (X=2 retr and Y=2 sec.) the child is suspended for a long interval: 2*2*2*3=24 seconds. Is it possible to avoid that chlidren remain suspended? Is there any configuration param that can reduce this interval? Is it possibile to introduce a little change in sip_resolvehost (check errno) in order to avoid the SRV and A queries if the NAPTR fails because the DNS servers are unreachable? ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

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[Kamailio-Users] Doubt about nat=yes
by mayamatakeshi 25 Jan '09

25 Jan '09

Hello, I'm trying to understand this sample cfg: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy This cfg is used as foundation for some of our cfg files here but nobody is sure about exactly how some things work. So I'm trying to clear things up. In particular, I am clueless about why there is code removing "nat=yes" from the URI and adding it to the Contact header (actually I don't know what is its meaning. I suppose this is a way of a client advertising that it is behind NAT, but I couldn't find which RFC defines this): subst_uri('/(sip:.*);nat=yes/\1/') search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); Is this something that always must be done when dealing with nat or it was a particular situation that the writer of the cfg had to workaround? I'm rewriting our cfg file (to be used with m4) and I'm trying to remove unnecessary things. Another thing in the cfg: in case a reinvite is issued, should not unforce_rtp_proxy be called before force_rtp_proxy is called again? Could this lead to having rtpproxy leaking resources? <http://www.kamailio.org/docs/modules/1.3.x/nathelper.html#AEN367> regards, mayama

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[Kamailio-Users] Openser-Asterisk Codec conversion..
by Rawshan Iajdani 24 Jan '09

24 Jan '09

My scenario is like that.. UA----->OpenSer(Outbound Proxy)---------Register Server | | | | Asterisk(codec converion)---------------------- The UA will register to Register server through outbound proxy OpenSer. When UA makes call it first comes to Openser, OpenSer should route the media to Register server through Asterisk for codec conversion. OpenSer will not hold any User account rather it will act as a proxy. The way I am thinking.. Openser could be used as outbound proxy that's fine. If it were used as outbound proxy, can it also route the media? In the asterisk side I may need username/password pair of UA to dial to Register Server in the second leg. Can I read the User/Pasword from Openser after UA is registered to OpenSer. I can have both OpenSer and Asterisk to reside on the same machine with different IPs so they can access the database if required..

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[Kamailio-Users] CarrierRoute With + Prefix
by Ali Jawad 23 Jan '09

23 Jan '09

Hi All I was able to setup carrierroute in database mode and when I make a call with prefix 00 it works fine. For that I did setup an entry in carrierroute, I did duplicate that same entry for prefix "+". If I call 004353453 it works fine however if I dial +12341234 I get a 403 forbidden error. The log shows: Jan 23 14:22:34 [25866] DBG:carrierroute:get_route_tree_by_id: tree default, domain 0 : 0 Jan 23 14:22:34 [25866] INFO:carrierroute:rewrite_uri_recursor: URI or route tree nodes empty, empty flag list Jan 23 14:22:34 [25866] INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri +9613041705, carrier 5, domain 0 Any guidance please ? With Regards

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[Kamailio-Users] [presence] use case of a user that doesn't un-REGISTER and un-PUBLISH
by Pascal Maugeri 23 Jan '09

23 Jan '09

Hi Using kamailio v1.5 (trunk), I'm testing the following simple presence scenario where one user shutdowns his application without prior un-PUBLISH and un-REGISTER. 1) user A subscribes to user B presence. Both use X-Lite. 2) user B shutdowns his X-Lite (without doing un-REGISTER and un-PUBLISH of his presence information), 3) user B starts again X-Lite, registers and publishes his presence, from that point when user B updates his presence information, user A does not see the updates ! I've configured kamailio to use save("location", "0x04") instead of save("location") for incoming REGISTER but this does not help. Is there a way to clean presence DB tables properly when user B register again ? Regards, Pascal

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[Kamailio-Users] Carrieroute help
by Ali Jawad 23 Jan '09

23 Jan '09

Hi All I was finally able to setup carrierroute on Solaris :). I am using carrierroute in db mode and I did setup two entries with rewritehost in carrierroute and two entries in route_tree with one route being default. I am still very new to carrierroute as a sample start I do have two astersisk servers that will act as pstn_gw and I want to split the traffic in half between both servers. Can anyone please provide sample instructions on what to put into the route{} statement in this scenario and any main ideas about route_tree and carrierroute ? Thanks.

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[Kamailio-Users] Siremis configuration error
by Juan Asencio 22 Jan '09

22 Jan '09

Hi Ramona, I have installed Siremis, but I get the following error: Unable to write in the session save path [/home/www/siremis/siremis/session] Fatal error: Call to a member function GetSessionContext() on a non-object in /home/www/siremis/openbiz/bin/BizSystem.php on line 9 Do you have any suggestions about how to fix this? Thanks in advance, Juan.-

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[Kamailio-Users] Carrierroute and AVP
by rod 22 Jan '09

22 Jan '09

Hi, I'd like to know if there is a way to retrieve the information from the carrierroute module in AVPs. Example: - a prefix has matched, and I'd like to get the field "description" associated with this route and add it in the SIP header - a prefix has matched, and I'd like to get the field "rewrite_host" associated with this route and use it with xlog Thanks, rod.

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[Kamailio-Users] htable dictionary attack example
by jh＠tutpro.com 22 Jan '09

22 Jan '09

htable module README has a dictionary attack limitation example. i'm afraid to "try it at home", because it seems to me that it in turn opens up a dos attack possibility: exhausting proxy shared memory by generating requests with random $au values. in order to avoid that, the script should include check if $au exists before adding it to dictionary. on am i missing something? -- juha

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[Kamailio-Users] BYE to the wrong IP
by Danny Fröberg 22 Jan '09

22 Jan '09

I need some hints as to what can be going on; Running: openser-1.2.2-tls Mobile 1 à PSTN à Provider SER à Local SER à Asterisk (Callforward) à Provider SER à PSTN à Mobile 2 When Mobile 2 hangs up, Mobile 1 does not get disconnected but hangs with dead air. Provider SER: Is load balanced using standards DNS round robin with one CNAME pointing to two IP's Our BYE goes to the wrong IP even though we use; modparam( "rr", "enable_full_lr", 1 ) ... fix_nated_contact(); force_rport(); if ( ! search( "^Content-Length:\ 0" ) ) { fix_nated_sdp( "3" ); }; t_on_reply( "1" ); record_route(); if ( loose_route() ) { route( 1 ); break; }; I have tested with check_via = yes to no avail. Do any of you have any idea what might be causing this? I know the provided information is sparse but thought I'd try the short descriptions of the problem first, if it's a known miss config. Many Regards Danny Froberg

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[Kamailio-Users] Nathelper Error
by George Lee 22 Jan '09

22 Jan '09

Hi, I have a SIP phone setup behind a router. When I placed a call from this phone, I got this error from OpenSER when the INVITE message was received. Jan 21 13:57:44 prod-ipser-10 /usr/sbin/openser[2100]: ERROR:nathelper:extract_body: message body has length zero Jan 21 13:57:44 prod-ipser-10 /usr/sbin/openser[2100]: ERROR:nathelper:fix_nated_sdp_f: cannot extract body from msg! Jan 21 13:57:45 prod-ipser-10 /usr/sbin/openser[2097]: ERROR:nathelper:extract_body: message body has length zero Jan 21 13:57:45 prod-ipser-10 /usr/sbin/openser[2097]: ERROR:nathelper:fix_nated_sdp_f: cannot extract body fr Any ideas? Lee

3 2

Re: [Kamailio-Users] [Kamailio-Devel] SIREMIS - Web Management Interface for Kamailio (OpenSER)
by Elena-Ramona Modroiu 21 Jan '09

21 Jan '09

Victor Pascual Ávila wrote: > On Wed, Jan 21, 2009 at 5:45 PM, Iñaki Baz Castillo <ibc(a)aliax.net> wrote: > >> 2009/1/21 Elena-Ramona Modroiu <ramona(a)asipto.com>: >> >>> For few days, a demo site can be accessed at: >>> - http://siremis.asipto.com/demo/ >>> - username: admin and password: admin >>> >> Guau! really great! >> >> > > Great work. Thanks! > Thanks for your appreciations! Regards, Ramona

1 0

[Kamailio-Users] new transformation class: tobody
by Daniel-Constantin Mierla 21 Jan '09

21 Jan '09

Hello, trunk has a new transformation class: tobody The name comes from To header Body and provides access to fields inside strings that follow the grammar of a To header - such headers are From, P-Asserted-Identity, .... Real benefit of using this transformation is that it copes with strings that can have values either simple URIs or NameAddr. The format of transformation is {tobody.param}, where param can be: - display - return display name - uri - return uri - user - return username from uri - host - return hostname from uri - tag - return the value of tag parameter Documentation is available at: http://www.kamailio.org/dokuwiki/doku.php/transformations:devel#to-body_tra… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Error when fetching current user list
by Ali Jawad 21 Jan '09

21 Jan '09

Hi Guys At times when I want to get the current user list using /usr/local/sbin/openserctl ul show I get this error ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. I dont think this is a configuration error, as killing and restarting openser does fix this. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net <outbind://147/www.splendor.net> ) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] SIREMIS - Web Management Interface for Kamailio (OpenSER)
by Elena-Ramona Modroiu 21 Jan '09

21 Jan '09

Hi everybody, I want to announce the launch of SIREMIS Project, aiming to build web management interface for Kamailio (OpenSER). Later will follow also the sip-router.org project. The project was developed so far by Asipto - http://www.asipto.com - using OpenBiz PHP framework - the target was to make it easier to extend for non-web developers. Soon, the project sources will be hosted on sourceforge.net SVN (project name: siremis). Homepage of the project is: - http://siremis.asipto.com The first beta release is available for download and includes the features: - subscriber, database aliases and speed dial management - location table view - dispatcher, prefix-domain translation and least cost routing (lcr) management - access control lists (user groups) and permissions management - accounting records, missed calls and call data records view - hash table, dial plan table and user preferences table management - off-line message storage view - it was tested with MySQL but should support other sql-like DBs Internal design: * developed on top of a framework: PHP/Web2.0/Ajax * XML Metadata driven extensions * Model-View-Controller (MVC) architecture You can see some screenshots at: - http://siremis.asipto.com/photos/siremis-screenshots/ For few days, a demo site can be accessed at: - http://siremis.asipto.com/demo/ - username: admin and password: admin Download and installation details: - http://siremis.asipto.com/install/ I hope you will enjoy it and help to extend it. The plan is to release first stable version more or less at the same time with Kamailio (OpenSER) 1.5.0. Your feedback is welcome! Best regards, Ramona

1 0

Re: [Kamailio-Users] Problem With Carrierroute on Solaris
by Ali Jawad 21 Jan '09

21 Jan '09

It worked finally, As per hint I did change endif DEFS+=$(CONFUSEDEFS) LIBS=$(CONFUSELIBS) To endif CONFUSEDEFS=-I$(LOCALBASE)/include -I/usr/local/include -I/opt/include -I/usr/sfw/include CONFUSELIBS=-L$(LOCALBASE)/lib -L/usr/local/lib -L/usr/sfw/lib -L/opt/lib -lconfuse DEFS+=$(CONFUSEDEFS) LIBS=$(CONFUSELIBS) Now I got an error that Jan 21 14:36:24 [24863] INFO:carrierroute:bind_data_loader: use file as configuration sourceJan 21 14:36:24 [24863] ERROR:carrierroute:bind_data_loader: can't stat config file But this is a totally different story :), let me hit the carrierroute documentation. Thanks A million -----Original Message----- From: Henning Westerholt [mailto:henning.westerholt@1und1.de] Sent: 2009-01-21 15:32 To: Ali Jawad Subject: Re: [Kamailio-Users] Problem With Carrierroute on Solaris On Wednesday 21 January 2009, you wrote: > At current it is > > /opt/lib -lconfuse <-- note the L > > should I make that > > /opt/lib -confuse -lgcc > > or > > /opt/lib -lconfuse -lgcc <- double L The last one, with two 'l' is correct. The '-l' tells the compiler that this is a linker configuration, after the l comes the library name. '-L' is a library location.

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[Kamailio-Users] Job description
by laurent schweizer 21 Jan '09

21 Jan '09

JOB TITLE: VOIP SOFTWARE DEVELOPER LOCATION: Zurich/Switzerland JOB TYPE: Full time COMPANY: Peoplefone AG is a Pan-European VoIP provider present in 4 countries and headquartered in Zurich/Switzerland ( www.peoplefone.com). JOB DESCRIPTION: - Manage together with CEO/CTO the geographical expansion on IT level - Develop and maintain the VoIP Platform (OPENSER, billing software,…) - Create new VoIP features for the clients - Ensure excellent quality of VoIP service and appropriate security level - Supervise interactions with webmasters for the website development - Provide technicians and customer service of affiliates with technical support REQUIREMENTS: - Graduation from college or university with a Bachelor's degree(preferably IT) - Practical knowledge of C, C++, PHP and JAVA - Practical knowledge of database like Mysql and Postgresql - Linux experience - Experience with tools for Network analysis like Ethereal - VoIP knowledge with experience on OPENSER/ASTERISK platform - Fluency in English - Ability to interact with individuals and groups at all levels - Detail oriented and analytical - Strong verbal and written communication skills +41 44 553 20 01 (Office CH) +43 1 266 00 50 (Office A) +48 22 311 78 35 (Offfice PL) Alfred Escher-Strasse 26 Pottendorfer Str. 25-27 Al. Jerozolimskie 179 8002 Zurich/Switzerland 1120 Wien/Austria 02-222 Warsaw/Poland www.peoplefone.ch www.peoplefone.at * www.plfon.pl*

1 0

[Kamailio-Users] SEMS 1.1.0 released
by Stefan Sayer 20 Jan '09

20 Jan '09

Hello, Version 1.1.0 of SEMS, the SIP Express Media Server, has been released. SEMS is a free, high performance, extensible media and application server for SIP (RFC3261) based VoIP services. New in version 1.1 is: * DSM state machine scripting (it's cool!) * an (experimental) ISDN gateway module * binrpc: MT (SER->) and connection pool (->SER) * MT xmlrpc server * controlled server shutdown * improved logging * g722 in 8khz compat mode * out of dialog request handling for modules & dialogs without sessions * audio file autorewind, AmAudio mixing * SIP and media IP separately configurable * UID/DID support for voicemail/-box/annrecorder * and quite some bugs and mem leaks fixed, documentation, etc. You can get the sources from ftp://ftp.iptel.org/pub/sems/sems-1.1.0.tar.gz and find documentation and packages at ftp://ftp.iptel.org/pub/sems/1.1/1.1.0/ and a debian repository (etch/lenny) at deb http://ftp.iptel.org/pub/sems/debian etch free deb-src http://ftp.iptel.org/pub/sems/debian etch free Further documentation, links to lists, tracker etc is available at the project's homepage: http://iptel.org/sems/ I would like to thank for all contributions and patches, this time from Stefan Sayer, Raphael Coeffic, Grzegorz Stanislawski, Bogdan Pintea, Greger Teigre, Rui Jin Zheng, Alfred E Heggestad, Juha Heinanen, Peter Lemenkov, Peter Loeppky, Robert Szokovacs, Jeremy A, Alex Gradinar Apologies if you receive multiple copies of this message. Best Regards Stefan Sayer -- Stefan Sayer VoIP services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

2 2

[Kamailio-Users] URIs in "P-Asserted-Identity" headers
by Alex R.S.M 20 Jan '09

20 Jan '09

Hi, How can I extract the URIs in "P-Asserted-Identity" when there are multiple "P-Asserted-Identity" headers. Variable $ai return the URI in the first "P-Asserted-Identity" header. Thanks, Alex _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012…

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[Kamailio-Users] Problem With Carrierroute on Solaris
by Ali Jawad 20 Jan '09

20 Jan '09

Hi All I did remove carrierroute from the excluded modules and I did recompile, all went smooth however when I try to run the server with the module loaded I get this error : fatal: relocation error: file carrierroute.so: symbol cfg_init: referenced symbol not found echo $LD_LIBRARY_PATH /opt/csw/mysql5/lib/mysql/:/opt/csw/lib/:/opt/csw/include/:/opt/csw/incl ude/openssl/:/usr/local/lib/ find /usr/ -name "*confuse*" /usr/local/include/confuse.h /usr/local/lib/libconfuse.a /usr/local/lib/libconfuse.la /usr/local/lib/libconfuse.so /usr/local/lib/libconfuse.so.0 /usr/local/lib/libconfuse.so.0.0.0 Please advice.

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[Kamailio-Users] ERROR:tm:t_forward_nonack: no branch for forwarding
by Juan Asencio 20 Jan '09

20 Jan '09

Hi list, Has anyone a suggestion to solve this error or where it might come from? I'm using carrierroute module to route calls through to different domains, but I can't make it work properly. [root@localhost kamailio]# Jan 20 12:05:52 localhost /sbin/kamailio[24686]: INFO:carrierroute:cr_do_route: uri 123456 was rewritten to sip:123456@200.100.200.100 Jan 20 12:05:52 localhost /sbin/kamailio[24686]: ERROR:tm:t_forward_nonack: no branch for forwarding Jan 20 12:05:52 localhost /sbin/kamailio[24686]: ERROR:tm:w_t_relay: t_forward_nonack failed Jan 20 12:05:52 localhost /sbin/kamailio[24680]: CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 484 Thanks in advance. Juan.-

2 1

[Kamailio-Users] new comparison functions in config
by Daniel-Constantin Mierla 19 Jan '09

19 Jan '09

Hello, following discussions on the mailing lists and tracker related to string and URI/AoR comparison, textops and siputils introduce new dedicated functions: - cmp_str(str1, str2) - compare two strings case sensitive - cmp_istr(str1, str2) - compare two strings case insensitive - cmp_uri(str1, str2) - compare two URIs as per RFC3261(up to parameters comparison rules) - cmp_aor(str1, str2) - compare two address of records http://kamailio.org/docs/modules/devel/textops.html http://kamailio.org/docs/modules/devel/siputils.html Just to refresh, the string comparison using config operators (==, <=, ...) are case insensitive (== is equivalent to cmp_istr(...)). Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Vrrp/Linux-HA, paid consulting opportunity
by sergejf 19 Jan '09

19 Jan '09

Hello, I hope this is the right forum to post a request for consulting services. We have a requirement for setting up a HA OpenSIPS 1.4.2 installation with a primary (active) and secondary (failover) server running CentOS 5.x. SIP messages from the active server must come from a virtual IP address and carrier gateways receiving our call traffic must remain oblivious to any server failure. Failover must occur in < 5 seconds. We would like to see a simple solution based on open source components like vrrp or Linux-HA, including the SIP heartbeat mechanism and failover per se, as well as the recovery process once the primary server recovers from the failure causes. The work will be done in conjunction with our Linux Administrator and IP Telephony Engineer and must be documented, at least in a basic fashion. Failover testing under load will be performed prior to project acceptance. If you are an expert in OpenSER HA implementations and are interested to help us, please respond via this forum or directly to my e-mail address, serge A T elasticall D O T com. Thanks in advance. Best Regards, Serge -- View this message in context: http://www.nabble.com/Vrrp-Linux-HA%2C-paid-consulting-opportunity-tp215460… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] t_check_trans() fails
by Alex R.S.M 19 Jan '09

19 Jan '09

Hi, What might be the cause that t_check_trans() function fails and ACK message is not being routed? This is a simple SIP to SIP call. if ( is_method("ACK") ) { if ( t_check_trans() ) { # non 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.\n"); exit; } } _________________________________________________________________ Windows Live™ Hotmail®: Chat. Store. Share. Do more with mail. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_explor…

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[Kamailio-Users] Run Server in background with maximum debug to log file
by Ali Jawad 19 Jan '09

19 Jan '09

Hi I am running the server with debug=9 and fork=yes and log standard output = yes. when I do that I can see a lot of debug info and sip packets in the standard output however the log file does show only some notices and xlog entries. So when I close putty and get back later or a few days later I have to kill and restart openser to see debug info as the log file is not showing maximum debug levels unlike the standard output. Please advice, the goal I want to reach is to see maximum loggin in a file so I can check it out without having to restart the server. Thanks

2 1

[Serusers] [somewhat OT] seeking ideas/input for my thesis
by sp4rc 19 Jan '09

19 Jan '09

Hello VoIP guys Sorry for being somewhat off-topic. At the moment I am studying informatics in the seventh semester and I need to start thinking about my thesis. As I am very interested in VoIP technologies I thought about picking this as my main topic. So far I have only little experience in this area. I have been fiddling around with siproxd and pfSense and have red the one or the other packet dump containing SIP and RTP traffic, had a look into codecs, STUN, etc... but very cursorily, and that's the reason why I am quite unsure on which track to go. I think I am quite familiar with many network protocols and devices... so here comes the question of the questions: What would be a great project for my thesis to work on in the VoIP field? What are topics that still need special development? The time frame should be around 300 hours but don't take this value too seriously... Any ideas are welcome! /sp4rc

1 0

[Kamailio-Users] SNMPStats per request statistics, is that possible?
by Noel R. Morais 19 Jan '09

19 Jan '09

Hi Guys, Is possible to have per request statistics from the SNMPStats module? What I mean is if I can know how many INVITES, REGISTERS, SUBSCRIBERS, NOTIFIES and so on were processed by kamailio. is that possible? Cheers, Noel

2 1

[Kamailio-Users] new feature: run route blocks on timer
by Daniel-Constantin Mierla 17 Jan '09

17 Jan '09

Hello, a new module was committed in trunk, to be released as 1.5.0 - the plan at: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x The rtimer module allows to execute route block on timer. It can create new processes that will do the timer execution of the routes or use existing timer. There can be multiple timers and each timer can execute multiple route blocks.See readme at: http://kamailio.org/docs/modules/devel/rtimer.html Such features is useful if you want to do periodic jobs in your service 1) clean a database were you store temporary data - e.g., delete entries older than 2 hours modparam("rtimer", "timer", "name=ta;interval=10;mode=1;") modparam("rtimer", "exec", "timer=ta;route=8") route[8] { sql_query("ca", "delete from tmp where timeval<$TS-3600","ra"); } 2) save statistics in database so you can build graph, etc... http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:devel#statistics route[8] { sql_query("ca", "insert into kamailio_stats values ($stat(...), ...)","ra"); } 3) Along with another pretty new feature - sending sip requests from config file: http://lists.kamailio.org/pipermail/users/2009-January/021426.html you can implement keep-alive/syncronization mechanisms within your farm of sip routers. Any other ideas? Let's see the best one for using the new stuff from uac and the rtimer module... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] new features in htable module
by Elena-Ramona Modroiu 16 Jan '09

16 Jan '09

Hi, htable module has been enhanced with couple of features that give more flexibility in working as config file cache system. - auto-expire - to each value stored in hash table can be assigned an expire value. If no update was done in that time interval, the value will be automatically removed. One new pseudo-variable class has been added to manage the expire values: $shtex(htable=>key) - it can be used to get/set the time-out interval. To get this feature on, the hash table definition must include autoexpire attribute greater than 0. - sht_rm_name_re(htable=>regexp) - delete from hash table all entries that match the name against regexp - sht_rm_value_re(htable=>regexp) - delete from hash table all entries that match the value (string) against regexp - reload the content of a hash table from database at runtime upon MI command. The hash table definition must include the database table name, then you can issue: 'kamctl fifo sht_reload htname' to reload the hash table with name 'htname'. Check the documentation at: http://kamailio.org/docs/modules/devel/htable.html Regards, Ramona

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[Kamailio-Users] Registered user
by michel freiha 16 Jan '09

16 Jan '09

Dear All, I need to ask please about which function should I use in order to check while making a call if the user who is dialing the number is making the call from a registered account or not? Regards

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[Kamailio-Users] new feature: send sip requests from config file
by Daniel-Constantin Mierla 16 Jan '09

16 Jan '09

Hello, uac module provides a new function that allow sending sip requests from configuration file: - uac_send_req() http://kamailio.org/docs/modules/devel/uac.html#id2467912 It opens the door for a new bunch of things you can do now. I can think of now: 1) send IM alerts on special events route { $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:alert@kamailio.org"; $uac_req(furi)="sip:server@kamailio.org"; $uac_req(hdrs)="Content-Type: text/plain\r\n"; $uac_req(body)="SIP request from:" + $si + ":" + $sp; uac_send_req(); } 2) turn lamps on for some sip phones on some events - e.g., snom: http://wiki.snom.com/Features/LED_Remote_Control 3) talk server-to-server - here one example can be building a platform with a central server for accounting Accounting is done at 200ok. So each routing server will build a csv list (time,callid, from tag, to tag, invite from uri, invite r-uri): onreply_route[1] { if(status!="200") return; $uac_req(method)="ACCOUNTING"; $uac_req(ruri)="sip:store@accounting.kamailio.org"; $uac_req(furi)="sip:server@server1.kamailio.org"; $uac_req(hdrs)="Content-Type: text/accounting-csv\r\n"; pv_printf($uac_req(body), "$TS,$ci,$ft,$tt,$T_req($fu),$T_req($ru)"); uac_send_req(); } note that $T_req() is given by tm module, also with Kamailio trunk: http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:devel#tm_module_p… The accounting server: route{ # you should check the source IP also if(method=="ACCOUNTING" && $rU="store") { sql_query("ca", "insert into accounting(timeval,callid,ftag,ttag,src,dst) values ('$(rb{s.select,0,,})','$(rb{s.select,1,,})', ...)", "ra"); send_reply("200", "Stored"); } } ... and put your imagination work ... there are plenty of new and cute features you can add on your service with this functionality ... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] TCP ZeroWindow and TCP Window Full
by Gawith 16 Jan '09

16 Jan '09

Hi, I see enormous TCP problems on my systems. All of a sudden, a perfectly working TCP session uses ZeroWindow packets, later I see TCP Full Window packets and finally the peer resets the connection. In the logs I see a lot of "core:handle_io: empty fd map" messages. Has anyone experienced similar problems? Regards.

2 3

[Kamailio-Users] SIP Digest Access Authentication RELAY survey
by Victor Pascual Ávila 16 Jan '09

16 Jan '09

Hi, excuse me if this message is not directly related to Kamailio. I'm just wondering if folks could share with me if (and how) they have prevented the "SIP Digest Access Authentication RELAY" in their networks (and what worked for them or not). NAT boxes reduce dramatically the scenarios for a successful attack. Otherwise, some might be mitigating the attack by means of forcing UAs to use outbound proxies while others might be reducing the attack incentives by means of message integrity. Any comment would be appreciated, -- Victor Pascual Ávila

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[Kamailio-Users] WeSIP sans Tomcat.
by Alex Balashov 16 Jan '09

16 Jan '09

Will WeSIP run in Jetty? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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Re: [Kamailio-Users] serMyAdmin Install
by Flavio Goncalves 15 Jan '09

15 Jan '09

Please, post this questions also in the serMyAdmin forum www.sermyadmin.org. I think the main developer Leonardo is not a subscriber of the list. Flavio

1 0

Re: [Kamailio-Users] serMyAdmin
by Flavio Goncalves 15 Jan '09

15 Jan '09

Date: Thu, 15 Jan 2009 17:20:29 +0100 (CET) From: "Juan Asencio" <juan(a)asencio.dk> Subject: [Kamailio-Users] serMyAdmin To: "users" <users(a)lists.kamailio.org> Message-ID: <d077ccb2f33e62d0814f1b519fb7ebfb.squirrel(a)webmail1.b-one.net> Content-Type: text/plain;charset=iso-8859-1 Hi list, I would like to ask if serMyAdmin works with Kamailio and if it does what changes would be necessary to make it work?Cheers, Juan.->Yes it works. Just create a symbolic link. /etc/opensips/ln -s opensipsctl kamctl. Sorry about this. In the next wersion we are planning to provide a configuration fileto avoid this ugly solution. We want to support as many versions as feasible. Kamailio, OpenSIPs and OpenSER.Check www.sermyadmin.org for the latest version for OpenSER. Regards, Flavio E. Goncalves

1 0

[Kamailio-Users] sermyadmin install
by TCB 15 Jan '09

15 Jan '09

I hope this is the right list, i'm trying to run latest sermyadmin against a remote mysql DB. I get error of this nature. Last packet sent to the server was 0 ms ago. at java.security.AccessController.doPrivileged(Native Method) ... 22 more Caused by: org.springframework.jdbc.support.MetaDataAccessException: Could not get Connection for extracting meta data; nested exception is org.springframework.jdbc.CannotGetJdbcConnectionException: Could not get JDBC Connection; nested exception is com.mysql.jdbc.exceptions.jdbc4.CommunicationsException: Communications link failure -- TC

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[Kamailio-Users] [Fwd: RE: new developer: Inaki Baz Castillo]
by Daniel-Constantin Mierla 15 Jan '09

15 Jan '09

... probably was meant here ... -------- Original Message -------- Subject: RE: [Kamailio-Users] new developer: Inaki Baz Castillo Date: Thu, 15 Jan 2009 14:43:42 -0500 From: David A Céspedes <ingdavidcespedes(a)cable.net.co> To: 'Daniel-Constantin Mierla' <miconda(a)gmail.com> References: <496E1A35.9050605(a)gmail.com> Congratulations to Iñaki. He has help me a lot in my bachelor thesis about SIPconnect. Great add to the developers team. David Cespedes -----Mensaje original----- De: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] En nombre de Daniel-Constantin Mierla Enviado el: Miércoles, 14 de Enero de 2009 12:01 p.m. Para: kamailio; devel Asunto: [Kamailio-Users] new developer: Inaki Baz Castillo Hello, here it is the proper welcome of Iñaki Baz Castillo as developer -- SourceForge id: ibc_sf. Log time community member and active contributor with troubleshooting, patches and support emails, Inaki committed his first module today (announcements sent hours ago): http://lists.kamailio.org/pipermail/devel/2009-January/017544.html He has an extensive knowledge of SIP and he is know as PITA for those writing confusing RFCs. Thanks for your work and looking forward to collaborating in the project. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] disable simultaneous register
by Luis Guaman 15 Jan '09

15 Jan '09

Hi all, Does anybody let me know how to disable simultaneous registers, using the same user login/password in two or more different machines? thanks in advance Luis Guaman Interlancompu

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[Kamailio-Users] Running the server on 64 bit platform
by Ali Jawad 15 Jan '09

15 Jan '09

If I do download the latest version of the server and compile from source, I do have a sparc system with 64 bit. Will the server software run on 64bit ? With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net <outbind://41/www.splendor.net> ) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 1

[Kamailio-Users] CDRTool
by Brian Chamberlain 15 Jan '09

15 Jan '09

Hi All, Using CDRTool 6.7.1 with Kamailio 1.4.1, have the rating engine running now fine. Doing my accounting using Freeradius and the CDR's are showing up fine in CDRtool. I am starting to work on the rates and destinations etc.. One issue I may have, one of my providers makes me put *1001* in front of the number I am dialling and then the local code, not the international one.. How will I normalise this *1001* so that CDRTool will just read in the prefix? Also can anyone point me in the direction of good documentation for the rating engine? The rating.txt is a bit vague at best. TIA. Brian

1 0

Re: [Kamailio-Users] debian packages not found
by Henning Westerholt 15 Jan '09

15 Jan '09

On Thursday 15 January 2009, Amit Agarwal wrote: > Hello Henning, > Thanks, > But It was also my mistake - I didnt go through the document properly. > I managed to download the packages for debian from > http://www.kamailio.org/pub/kamailio/latest-1.4.x/packages/debian/etch/ Hi Amit, ok. Please keep in mind that this is not the latest release of the stable branch 1.4, so if you want to use kamailio in production, i'd suggest to use 1.4.3. It contains many bug fixes over the package you use now. Cheers, Henning

1 0

[Kamailio-Users] new: simplified compile options system
by Daniel-Constantin Mierla 15 Jan '09

15 Jan '09

Hello, trunk includes now an easier system to give the compile options. So far, changing the size of private memory, number of routes that can be used in config file required edit of source code. Other required changes in different files, so we looked to simplify and unify the mode of setting most common options. A new file, Makefile.vars holds now options in format name=value where you can turn off/on compilation flags (like compile in memory debug mode), tune internal values and specify what group of modules (based on dependency) to be compile as well. So, from now on, if you want to tune compilation parameters, check first the Makefile.vars file. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] debian packages not found
by Amit Agarwal 15 Jan '09

15 Jan '09

Hi, I have looked into the following paths, But I could not find the required package. http://www.kamailio.org/debian/dists/stable/main/binary-i386/ http://www.kamailio.org/debian/dists/stable/non-free/binary-i386/ http://www.kamailio.org/debian/dists/etch/main/binary-i386/ http://www.kamailio.org/debian/dists/etch/non-free/binary-i386/ Also, I wish to know the difference in the packages contained in the above folders. Thanks, Regards, Amit

2 1

[Kamailio-Users] new developer: Inaki Baz Castillo
by Daniel-Constantin Mierla 14 Jan '09

14 Jan '09

Hello, here it is the proper welcome of Iñaki Baz Castillo as developer -- SourceForge id: ibc_sf. Log time community member and active contributor with troubleshooting, patches and support emails, Inaki committed his first module today (announcements sent hours ago): http://lists.kamailio.org/pipermail/devel/2009-January/017544.html He has an extensive knowledge of SIP and he is know as PITA for those writing confusing RFCs. Thanks for your work and looking forward to collaborating in the project. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 2

[Kamailio-Users] New module "Regex"
by Iñaki Baz Castillo 14 Jan '09

14 Jan '09

Hi all, I announce the first commit of "Regex" module to Kamailio trunk. This module offers matching operations against regular expressions using the powerful PCRE library. A text file containing regular expressions categorized in groups is compiled when the module is loaded, storing the compiled PCRE objects in an array. A function to match a string or pseudo-variable against any of these groups is provided. The text file can be modified and reloaded at any time via a MI command. The module also offers a function to perform a PCRE matching operation against a regular expression provided as function parameter. The module documentation is already online: http://kamailio.org/docs/modules/devel/regex.html The commit has been done in rev 5458: http://openser.svn.sourceforge.net/viewvc/openser?view=rev&revision=5458 Any suggestion is really welcome. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 2

[Kamailio-Users] snmpstats error
by Noel R. Morais 14 Jan '09

14 Jan '09

Hi guys, I'm trying to use the snmpstats module but on kamailio initialization it shows me the error: kamailio: ERROR:core:sr_load_module: could not open module </usr/local/lib/kamailio/modules/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefine d symbol: make_tempfile I've found in this lists a post asking to remove the follow line from Makefile: BUILDAGENTLIBS:=$(filter-out %netsnmpmibs, $(BUILDAGENTLIBS)) but this line doesn't exists. Does anyone know how to fix that? My system: Kamailio 1.4.2 notls CentOS 5 Kernel 2.6.18-92 Thanks in advance, Noel

3 4

[Kamailio-Users] perlvdb
by victorpy 13 Jan '09

13 Jan '09

Hello, i could be wrong but what i read from perlvdb is that using that module you abstract from the DB engine. I want to know if it's posible with this module to load a database in memory. I want to do this for performance. if is posible could you give an example, otherwise you could give a hint to solve this thanks bye -- View this message in context: http://www.nabble.com/perlvdb-tp21421051p21421051.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] playing different announcements to user
by Luciano Afranllie 13 Jan '09

13 Jan '09

Hi, I have not experience doing this so I want to ask you what is the best approach to instruct a media server about what announcement to play to the user. For example, if I use asterisk, should I define one extension per announcement type? Is this the common approach? Thanks Luciano

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Re: [Kamailio-Users] WARNING:core:send2child: no free tcp receiver
by Ali Jawad 13 Jan '09

13 Jan '09

Adding to the below I am getting these errors Jan 12 12:35:30 sero /usr/local/sbin/openser[30076]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jan 12 12:35:30 sero /usr/local/sbin/openser[30076]: ERROR:core:tcp_send: connect failed Jan 12 12:35:30 sero /usr/local/sbin/openser[30076]: ERROR:sl:msg_send: tcp_send failed Jan 12 12:35:30 sero /usr/local/sbin/openser[30076]: ERROR:auth:challenge: failed to send the response Jan 12 12:35:31 sero /usr/local/sbin/openser[30076]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Jan 12 12:35:31 sero /usr/local/sbin/openser[30076]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net <outbind://210/www.splendor.net> ) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554 ________________________________ From: Ali Jawad Sent: 2009-01-12 12:33 To: Ali Jawad; users(a)lists.opensips.org Cc: users(a)lists.kamailio.org Subject: RE: [OpenSIPS-Users] WARNING:core:send2child: no free tcp receiver As an update I have retrieved this [root@sero ~]# /usr/local/sbin/openserctl ps Process:: ID=0 PID=30033 Type=attendant Process:: ID=1 PID=30036 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=2 PID=30037 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=3 PID=30038 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=4 PID=30039 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=5 PID=30041 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=6 PID=30042 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=7 PID=30043 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=8 PID=30044 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=9 PID=30045 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=10 PID=30046 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=11 PID=30047 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=12 PID=30048 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=13 PID=30049 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=14 PID=30050 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=15 PID=30051 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=16 PID=30052 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=17 PID=30053 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=18 PID=30054 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=19 PID=30055 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=20 PID=30056 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=21 PID=30057 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=22 PID=30060 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=23 PID=30062 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=24 PID=30064 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=25 PID=30065 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=26 PID=30066 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=27 PID=30067 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=28 PID=30068 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=29 PID=30069 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=30 PID=30070 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=31 PID=30071 Type=timer Process:: ID=32 PID=30072 Type=timer Process:: ID=33 PID=30073 Type=MI FIFO Process:: ID=34 PID=30074 Type=TCP receiver Process:: ID=35 PID=30075 Type=TCP receiver Process:: ID=36 PID=30076 Type=TCP receiver Process:: ID=37 PID=30077 Type=TCP receiver Process:: ID=38 PID=30079 Type=TCP receiver Process:: ID=39 PID=30080 Type=TCP receiver Process:: ID=40 PID=30081 Type=TCP receiver Process:: ID=41 PID=30082 Type=TCP receiver Process:: ID=42 PID=30084 Type=TCP receiver Process:: ID=43 PID=30085 Type=TCP receiver Process:: ID=44 PID=30086 Type=TCP receiver Process:: ID=45 PID=30087 Type=TCP receiver Process:: ID=46 PID=30090 Type=TCP receiver Process:: ID=47 PID=30091 Type=TCP receiver Process:: ID=48 PID=30092 Type=TCP receiver Process:: ID=49 PID=30094 Type=TCP receiver Process:: ID=50 PID=30095 Type=TCP receiver Process:: ID=51 PID=30096 Type=TCP receiver Process:: ID=52 PID=30097 Type=TCP receiver Process:: ID=53 PID=30098 Type=TCP receiver Process:: ID=54 PID=30099 Type=TCP receiver Process:: ID=55 PID=30100 Type=TCP receiver Process:: ID=56 PID=30101 Type=TCP receiver Process:: ID=57 PID=30102 Type=TCP receiver Process:: ID=58 PID=30103 Type=TCP receiver Process:: ID=59 PID=30104 Type=TCP receiver Process:: ID=60 PID=30105 Type=TCP receiver Process:: ID=61 PID=30107 Type=TCP receiver Process:: ID=62 PID=30109 Type=TCP receiver Process:: ID=63 PID=30110 Type=TCP receiver Process:: ID=64 PID=30111 Type=TCP main With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net <outbind://206/www.splendor.net> ) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554 ________________________________ From: users-bounces(a)lists.opensips.org [mailto:users-bounces@lists.opensips.org] On Behalf Of Ali Jawad Sent: 2009-01-12 02:31 To: users(a)lists.opensips.org Cc: users(a)lists.kamailio.org Subject: [OpenSIPS-Users] WARNING:core:send2child: no free tcp receiver Hi All I am getting a lot of WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) I tried setting children to 20 that did not help, these warning start being displayed right after I start openser. Users frequently get service unavailable errors randomly when they try to connect, but I dont know if this is related to this warning message or not. As for the server there is plenty of CPU and RAM left, users range from 140 to 250 online all using tls. As for server limits I have set them to the following : You have new mail in /var/spool/mail/root [root@sero openser]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 101024 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16382 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Is this warning message related to the serice unavailable problem, what does the warning message mean in more depth and how to get rid of it ? Thanks Getting alot of Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10)

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Re: [Kamailio-Users] About Kamailio-1.4.0-notls.
by Sunil Teli 13 Jan '09

13 Jan '09

And also when I(Application A) send the Subscribe message with the Dialog event to the kamailio server, The kamailio server is not forwarding that Subscribe message to party B. Since kamailio acts as proxy it has to forward the Subscribe message to the party to whom I have Subscribed(ie Party B an x-lite phone). The call flow is as follows. REISTER A(APP) kamailio B(x-lite) C(x-lite) | | | | 200 OK | | | | | | | | REGISTER REGISTER | | | | | | | | 200 OK | | | | | | | | 200 OK | | | | | | | | SUBSCRIBE | | | | | | | | 202 OK | | | | | | | | NOTIFY | | | | Call Established between B and C | | | | 200 OK | | | | | | | | | | | | | | | | I am using kamailio-1.4.0-notls and trying to use dialog event package. The above call flow is as follows. 1) B an x-lite phone is Registered(B@kamailio) then C another x-lite phone is registered(c@kamailio) 2) Now from my application I am registering A and got registered(A@kamailio). 3) Now A is subscribing to B for the dialog event, so I am sending SUBSCRIBE message to kamailio server, that is I am sending SUBSCRIBE message from A@application to B@kamailio with the Event header. 4) Now I got 202 for the SUBSCRIBE. 5) Server Kamailio sends NOTIFY and then my application A sends 200 OK. 6) Now I will establish call between two x-lite phones B and C. Now My Question? When I establish call between B and C, I(Application A) have to get NOTIFY from kamailio server about the dialog event, I am not getting the NOTIFY for the dialog event since I have Subscribed to B for the Dialog event Package. Finally I(A application) want to know that dialog changes in the B. But Dialog module is present in the kamailio code. So for what purpose is it used? Is there any modules I have to include or provision in the config file or make file, If yes then where I have to make the changes. Thank You Regards, Sunil Teli -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Monday, January 12, 2009 7:08 PM To: Sunil Teli Cc: users Subject: Re: [Kamailio-Users] About Kamailio-1.4.0-notls. Hi Sunil! Kamailio 1.4 does not support the "dialog" event package. You have to use Kamailio "trunk" version. Trunk includes 2 new modules: PUA_DIALOGINFO: generates PUBLISH requests to publish the dialog state PRESENCE_DIALOGINFO: processes PUBLISH requests, aggregate dialog information and send it with NOTIFY to the watcher Please see the detailed READMEs of these modules. http://www.kamailio.org/docs/modules/devel/ regards klaus ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahindra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahindra.com/Disclaimer.html</a> internally within Tech Mahindra. ============================================================================================================================

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[Serusers] db_query in failure_route
by John G. Rose 12 Jan '09

12 Jan '09

I noticed that a db_query in a failure route works in some situations but not in others. Is there way to make it work all the time? I know in SER <2.0 you are supposed to not do this. In 2.0 it works sometimes. Also I noticed in some db_ops documentation there is the db_proper() failure route hack fix but this is not in SER 2.0 db_ops code apparently... John

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[Kamailio-Users] Segment fault with trunk version and default config file
by Iñaki Baz Castillo 12 Jan '09

12 Jan '09

Hi, I've upgraded Kamailio to the trunk version (ref 5440). i've compiled in with "make deb-etch" in my Debian Etch 64 bits as so many times. If I start Kamailio "with /etc/init.d/kamailio start" I get the following logs in which no error appear, but are obviously incomplete: ------------------------- Not starting kamailio: invalid configuration file! Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/sl.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/tm.so Jan 12 18:10:02 [8915] DBG:core:register_module: register_pv: tm Jan 12 18:10:02 [8915] DBG:core:pv_table_add: PV table not initialized, doing it now Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/rr.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/pv.so Jan 12 18:10:02 [8915] DBG:core:tr_table_add: TR table not initialized, doing it now Jan 12 18:10:02 [8915] DBG:core:register_module: register_pv: pv Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/maxfwd.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/usrloc.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/registrar.so Jan 12 18:10:02 [8915] DBG:core:register_module: register_pv: registrar Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/textops.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/mi_fifo.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/uri_db.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/siputils.so Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/xlog.so Jan 12 18:10:02 [8915] DBG:core:register_module: register_pv: xlog Jan 12 18:10:02 [8915] DBG:core:yyparse: loading module /usr/lib/kamailio/modules/acc.so Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: mi_fifo matches module mi_fifo Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <fifo_name> in module mi_fifo [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: rr matches module rr Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <enable_full_lr> in module rr [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: rr matches module rr Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <append_fromtag> in module rr [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: registrar matches module registrar Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <method_filtering> in module registrar [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: uri_db matches module uri_db Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <use_uri_table> in module uri_db [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: uri_db matches module uri_db Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <db_url> in module uri_db [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <early_media> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <report_ack> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <report_cancels> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <detect_direction> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <failed_transaction_flag> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <log_flag> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <log_missed_flag> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <db_flag> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: acc matches module acc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <db_missed_flag> in module acc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: usrloc matches module usrloc Jan 12 18:10:02 [8915] DBG:core:set_mod_param_regex: found <db_mode> in module usrloc [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <mf_process_maxfwd_header>(1) in module maxfwd [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <has_totag>(0) in module siputils [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <loose_route>(0) in module rr [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_check_trans>(0) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_relay>(0) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_check_trans>(0) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_relay>(0) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_check_trans>(0) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <record_route>(0) in module rr [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <append_hf>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <save>(1) in module registrar [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <sl_reply_error>(0) in module sl [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <lookup>(1) in module registrar [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_newtran>(0) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_reply>(2) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <sl_send_reply>(2) in module sl [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <is_method>(1) in module textops [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_on_branch>(1) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_on_reply>(1) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_on_failure>(1) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_relay>(0) in module tm [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <sl_reply_error>(0) in module sl [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <xlog>(1) in module xlog [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <xlog>(1) in module xlog [/usr/lib/kamailio/modules/] Jan 12 18:10:02 [8915] DBG:core:find_cmd_export_t: found <t_was_cancelled>(0) in module tm [/usr/lib/kamailio/modules/] ---------------------------------- If I run directly "kamailio" I get a segmentation fault, this explain that there is no error log above. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] About Kamailio-1.4.0-notls.
by Sunil Teli 12 Jan '09

12 Jan '09

Hi, I want to know that Kamailio-1.4.0-notls supports Dialog Event Package, If yes what configuration I have to do. If no the which version of the kamailio/OpenSER/OpenSIPS I have to use for the Dialog Event package support. Please go through the below call flow I have a problem. I am using kamailio-1.4.0-notls and trying to use dialog event package. The above call flow is as follows. 1) B an x-lite phone is Registered(B@kamailio) then C another x-lite phone is registered(c@kamailio) 2) Now from my application I am registering A and got registered(A@kamailio). 3) Now A is subscribing to B for the dialog event, so I am sending SUBSCRIBE message to kamailio server, that is I am sending SUBSCRIBE message from A@application to B@kamailio with the Event header. 4) Now I got 202 for the SUBSCRIBE. 5) Server Kamailio sends NOTIFY and then my application A sends 200 OK. 6) Now I will establish call between two x-lite phones B and C. Now My Question? When I establish call between B and C, I(Application A) have to get NOTIFY from kamailio server about the dialog event, I am not getting the NOTIFY for the dialog event since I have Subscribed to B for the Dialog event Package. Finally I(A application) want to know that dialog changes in the B Any help is appreciated to solve my problem. Advance thanks for the help. Thank You Regards, Sunil ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahindra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahindra.com/Disclaimer.html</a> internally within Tech Mahindra. ============================================================================================================================

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[Kamailio-Users] Kamailio (OpenSER) 1.5.0 Freezing on Jan 26, 2009
by Daniel-Constantin Mierla 12 Jan '09

12 Jan '09

Hello, we approach the time for a new stable version: http://www.kamailio.org/dokuwiki/doku.php/development:release-policy Proposed date to freeze the code is: Mon, Jan 26, 2009. It will follow about one month of testing and the the new version will be out. There is a lot of new development since last major release 1.4.0, check a summary at: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x AFAIK, before release, 2-4 new more module are going to be introduced and couple of existing modules extended. This is a notification for all users and developers (existing or upcoming) to finish the work before the freezing data if they want to get their code in Kamailio (OpenSER) 1.5.0. The 1.5.x series will be the last to use the current core. Next releases will be based on SIP Router project: http://sip-router.org/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] textops: regexp-based search/remove headers
by Daniel-Constantin Mierla 12 Jan '09

12 Jan '09

Hello, the textops module has two new functions that help to search and remove headers based on regular expressions: - is_present_hf_re(regexp) http://kamailio.org/docs/modules/devel/textops.html#id2506553 - remove_hf_re(regexp) http://kamailio.org/docs/modules/devel/textops.html#id2506863 Particularly the second is useful to remove a bunch of headers in one step. For example, if you have several proxyes in your platform and you exchange information between them via X-Something headers, then you can remove all with: remove_hf_re("^X-"); Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [Kamailio-Users] [OpenSIPS-Users] WARNING:core:send2child: no free tcp receiver
by Ali Jawad 12 Jan '09

12 Jan '09

As an update I have retrieved this [root@sero ~]# /usr/local/sbin/openserctl ps Process:: ID=0 PID=30033 Type=attendant Process:: ID=1 PID=30036 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=2 PID=30037 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=3 PID=30038 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=4 PID=30039 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=5 PID=30041 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=6 PID=30042 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=7 PID=30043 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=8 PID=30044 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=9 PID=30045 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=10 PID=30046 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=11 PID=30047 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=12 PID=30048 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=13 PID=30049 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=14 PID=30050 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=15 PID=30051 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=16 PID=30052 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=17 PID=30053 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=18 PID=30054 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=19 PID=30055 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=20 PID=30056 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=21 PID=30057 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=22 PID=30060 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=23 PID=30062 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=24 PID=30064 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=25 PID=30065 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=26 PID=30066 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=27 PID=30067 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=28 PID=30068 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=29 PID=30069 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=30 PID=30070 Type=SIP receiver udp:xx.yy.zz.aa:5060 Process:: ID=31 PID=30071 Type=timer Process:: ID=32 PID=30072 Type=timer Process:: ID=33 PID=30073 Type=MI FIFO Process:: ID=34 PID=30074 Type=TCP receiver Process:: ID=35 PID=30075 Type=TCP receiver Process:: ID=36 PID=30076 Type=TCP receiver Process:: ID=37 PID=30077 Type=TCP receiver Process:: ID=38 PID=30079 Type=TCP receiver Process:: ID=39 PID=30080 Type=TCP receiver Process:: ID=40 PID=30081 Type=TCP receiver Process:: ID=41 PID=30082 Type=TCP receiver Process:: ID=42 PID=30084 Type=TCP receiver Process:: ID=43 PID=30085 Type=TCP receiver Process:: ID=44 PID=30086 Type=TCP receiver Process:: ID=45 PID=30087 Type=TCP receiver Process:: ID=46 PID=30090 Type=TCP receiver Process:: ID=47 PID=30091 Type=TCP receiver Process:: ID=48 PID=30092 Type=TCP receiver Process:: ID=49 PID=30094 Type=TCP receiver Process:: ID=50 PID=30095 Type=TCP receiver Process:: ID=51 PID=30096 Type=TCP receiver Process:: ID=52 PID=30097 Type=TCP receiver Process:: ID=53 PID=30098 Type=TCP receiver Process:: ID=54 PID=30099 Type=TCP receiver Process:: ID=55 PID=30100 Type=TCP receiver Process:: ID=56 PID=30101 Type=TCP receiver Process:: ID=57 PID=30102 Type=TCP receiver Process:: ID=58 PID=30103 Type=TCP receiver Process:: ID=59 PID=30104 Type=TCP receiver Process:: ID=60 PID=30105 Type=TCP receiver Process:: ID=61 PID=30107 Type=TCP receiver Process:: ID=62 PID=30109 Type=TCP receiver Process:: ID=63 PID=30110 Type=TCP receiver Process:: ID=64 PID=30111 Type=TCP main With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net <outbind://206/www.splendor.net> ) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554 ________________________________ From: users-bounces(a)lists.opensips.org [mailto:users-bounces@lists.opensips.org] On Behalf Of Ali Jawad Sent: 2009-01-12 02:31 To: users(a)lists.opensips.org Cc: users(a)lists.kamailio.org Subject: [OpenSIPS-Users] WARNING:core:send2child: no free tcp receiver Hi All I am getting a lot of WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) I tried setting children to 20 that did not help, these warning start being displayed right after I start openser. Users frequently get service unavailable errors randomly when they try to connect, but I dont know if this is related to this warning message or not. As for the server there is plenty of CPU and RAM left, users range from 140 to 250 online all using tls. As for server limits I have set them to the following : You have new mail in /var/spool/mail/root [root@sero openser]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 101024 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16382 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Is this warning message related to the serice unavailable problem, what does the warning message mean in more depth and how to get rid of it ? Thanks Getting alot of Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10)

1 0

[Kamailio-Users] WARNING:core:send2child: no free tcp receiver
by Ali Jawad 12 Jan '09

12 Jan '09

Hi All I am getting a lot of WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) I tried setting children to 20 that did not help, these warning start being displayed right after I start openser. Users frequently get service unavailable errors randomly when they try to connect, but I dont know if this is related to this warning message or not. As for the server there is plenty of CPU and RAM left, users range from 140 to 250 online all using tls. As for server limits I have set them to the following : You have new mail in /var/spool/mail/root [root@sero openser]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 1024 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 101024 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 16382 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Is this warning message related to the serice unavailable problem, what does the warning message mean in more depth and how to get rid of it ? Thanks Getting alot of Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:32 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (7) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (8) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:33 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (9) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:34 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10) Jan 12 00:00:35 sero /usr/local/sbin/openser[28281]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (10)

1 0

[Serusers] ser behind NAT
by WANG Zhenzhong 11 Jan '09

11 Jan '09

Hi every body My SER is behind NAT, so when SER forward INVITE message, the VIA must be set to the public IP How? Thank you very much. 王振中 wangzz 艾诺通信系统（苏州）有限责任公司苏州工业园区国际科技园一期203A wangzz(a)avonaco.com ============================================

4 4

[Kamailio-Users] compare uri and aor
by Daniel-Constantin Mierla 11 Jan '09

11 Jan '09

Hello, I have introduced in siputils module functions to compare SIP URI and AoR in a more RFC manner, related to: https://sourceforge.net/tracker/index.php?func=detail&aid=2047019&group_id=… Now, for URI it does comparison of: - username - password - hostname - port For AoR, it does: - username - hostname - port (if port is missing, it assumes 5060 -- for URI comparison this is not used, conform to RFC3261) The questions are: - shall the comparison for URI be extended to follow full RFC? could get complex when taking URI headers in account -- haven't seen many, but the RFC allow them. Where to stop then the rules for comparing the URI? - for AoR, I am not sure if port should be considered, but when running multiple instances on different ports, it may make the difference? What do you think? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 5

[Serusers] SER monitoring capabilities
by Martin Lane 09 Jan '09

09 Jan '09

I'm looking for some information on the ability to monitor (have an indication of activity of, not listen too) the active calls real-time with the SER/SEMS suite. Can this be done and is anyone doing that? mll

3 2

[Serusers] SER monitoring capabilities
by Martin Lane 09 Jan '09

09 Jan '09

I'm looking for some information on the ability to monitor (have an indication of activity of not listen too) the active calls real-time with the SER/SEMS suite. Can this be done and is anyone doing that? mll

1 0

[Kamailio-Users] Help!! Getting get_hdr_field: out of memory
by Krunal Patel 09 Jan '09

09 Jan '09

Hi, I am using openser 1.2.3. I have increased PKG_MEM_POOL_SIZE to 4MB. Openser worked properly for 4 days since last restart. I found get_hdr_field: out of memory for all messages. I did openser restart & the issue is fixed now. Is anybody there who have faced the same issue? Thanks in advance!! -- Krunal patel

2 15

[Kamailio-Users] Internet drops and ATA don´t register
by MagoDaViola 09 Jan '09

09 Jan '09

Untitled Document Hello all .... I'm having a problem with our server or customers. Most of our clients access the server using the combination ATA and ADSL Modem, but when the ADSL drops and returns in a few minutes, the ATA can no longer make the register on SIP server. Sometimes passing through the modem or the requests of the ATA. I identified that this happens a lot with the D-Link modems DSL-500B and siemens SpeedStream 4200. If someone went through this I would your help ... thanks -------------------------------------------------------------------------- Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. a.. MSN: a_schleder(a)hotmail.com b.. Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular + 55 54 9966 7591 alcindo(a)processa.org Fone +55 54 3286-1738 comercial(a)processa.org

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