- sr-users - kamailio.org

[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

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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

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[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

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[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

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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*

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[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

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[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.-

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[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

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[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

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[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

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[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

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[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.

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[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

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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

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[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

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[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

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[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

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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

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[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

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[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

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[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

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[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>

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[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

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[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. ============================================================================================================================

2 1

[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

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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

9 34

[Kamailio-Users] Perl var main script access
by Luis Guaman 08 Jan '09

08 Jan '09

Hi , How can I get access in the onreply_route main script section for a variable set in a perl which is called from a previous route[] section?? Thanks in advance Luis Guaman Interlancompu

3 3

[Kamailio-Users] I need termine a call then timeout.
by igor 08 Jan '09

08 Jan '09

I need termine a call then timeout. but, return a message WARNING and do not Hangup! help me! WARNING:dialog:get_expired_dlgs: start with tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 (101) at 101 and end with end=0xb59cf550 end->prev=0xb5a11a20 end->next=0xb5a11a20 WARNING:dialog:get_expired_dlgs: getting tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 with 101 WARNING:dialog:get_expired_dlgs: end with tl=0xb59cf550 tl->prev=0xb5a11a20 tl->next=0xb5a11a20 and d_timer->first.next->prev=(nil) WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID 'T8x-xgYUoIe3707wDPg1qwq-AX4o3m' and tags '9697t1cm3thc6l2207g9' '7jju88u987' loadmodule "dialog.so" modparam("dialog", "enable_stats", 1) modparam("dialog", "db_url", "mysql://user:pass@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(s:dlgtimeout)") modparam("dialog", "dlg_extra_hdrs", "Hint: credit expired\r\n") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 10) if(is_method("INVITE")) { $avp(s:dlgtimeout)=5; $avp(s:ringtimeout)=10; setflag(4); route(4); } Atenciosamente, Igor Marques celular: 55 21 7858 7499 nextel: 87*24074 e-mail: igor(a)carneiro.cc web: www.carneiro.cc -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

3 7

[Kamailio-Users] Accounting: How to avoid a fraudulent BYE with lower CSeq?
by Iñaki Baz Castillo 08 Jan '09

08 Jan '09

Hi, first of all I'm sorry for this cross-posting, but I think it could be interesting for any SIP proxy with accounting capabilities. I'm thinking in the following flow in which the caller/attacker would get an unlimited call (but a limited CDR duration): -------------------------------------------------------------------------- attacker Kamailio (Acc) gateway INVITE (CSeq 12) ------> <-------- 407 Proxy Auth INVITE (CSeq 13) ------> INVITE (CSeq 13) ------> <------------------- 200 Ok <------------------- 200 Ok << Acc START >> ACK (CSeq 13) -----------> ACK (CSeq 13) -----------> <******************* RTP ************************> # Fraudulent BYE !!! BYE (CSeq 10) -----------> << Acc STOP >> BYE (CSeq 10) -----------> <-- 500 Req Out of Order <-- 500 Req Out of Order -------------------------------------------------------------------------- The call hasn't finished, but Kamailio has ended the accounting for this call since it received a BYE. And this BYE will generate a correct ACC Stop action (since it matches From_tag, To_tag and Call-ID). I think this is *VERY* dangerous and I hope I'm wrong. Would help the dialog module here? does the dialog module check the CSeq of the BYE in some way and could it prevent Kamailio from generating the ACC STOP action? (I don't think so). I've also asked in SIP-implementors and an idea could be generating the ACC STOP action when receiving the 200 OK for the BYE (and not when receiving the BYE itself). Of course this will be valid when the gateway is the recipient of the BYE (and we know the gateway is not an "attacker"), but this is not valid when the recipient of the BYE is an user since it could send no reply for the BYE. The only solution I see is: - Using the dialog module, Kamailio should check the CSeq value in the BYE. 1) Kamailio should forward the BYE just in case the CSeq is higher than the actual CSeq for this dialog direction. 2) Kamailio should generate the ACC STOP action just in case the CSeq is higher than the actual CSeq for this dialog direction. Both 1) and 2) are needed since a gateway could accept a BYE with wrong CSeq. In this case the call is ended but the accounting STOP action doesn't exist (infinite call). But I think this is too complex, isn't it? -- Iñaki Baz Castillo <ibc(a)aliax.net>

8 47

[Kamailio-Users] (no subject)
by igor 08 Jan '09

08 Jan '09

WARNING:dialog:get_expired_dlgs: start with tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 (101) at 101 and end with end=0xb59cf550 end->prev=0xb5a11a20 end->next=0xb5a11a20 Jan 8 11:52:22 voip01 /usr/sbin/kamailio[12596]: WARNING:dialog:get_expired_dlgs: getting tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 with 101 Jan 8 11:52:22 voip01 /usr/sbin/kamailio[12596]: WARNING:dialog:get_expired_dlgs: end with tl=0xb59cf550 tl->prev=0xb5a11a20 tl->next=0xb5a11a20 and d_timer->first.next->prev=(nil) Jan 8 11:52:22 voip01 /usr/sbin/kamailio[12596]: WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID 'T8x-xgYUoIe3707wDPg1qwq-AX4o3m' and tags '9697t1cm3thc6l2207g9' '7jju88u987' Atenciosamente, Igor Marques celular: 55 21 7858 7499 nextel: 87*24074 e-mail: igor(a)carneiro.cc web: www.carneiro.cc -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

1 0

[Serusers] SER server reload
by Leon Li 08 Jan '09

08 Jan '09

Dear all, I am new to SER and on a project to reload company's SER server for some SIP testing. However, I couldn't start SER after compiling. Every time I run "/usr/local/sbin/ser -f /PATH/ser.cfg", I got "ERROR: bad config file (1 errors)". Does this mean my ser.cfg has something wrong or where can I trace the logs for errors? Also, how do I uninstall the old SER and run a clean installation? Any suggestion would be appreciated. Best regards, Leon Leon Li | IT Engineer | Application & Services | AARNet Pty Ltd ________________________________ street address: Level 2, Binary Centre, Bldg1, 3 Richardson Place, North Ryde, NSW 2113 t. +61 2 9779 6937 m. 0403 603 473 e. leon.li(a)aarnet.edu.au w. www.aarnet.edu.au <http://www.aarnet.edu.au/> ________________________________ important This email and any files transmitted with it are confidential, and the rights of confidentiality in such information are not waived or lost by its mistaken delivery to you. Any dissemination, copying, use or disclosure of the email and/or such files without the permission of AARNet, or the sender, is strictly prohibited. If you have received this email in error, please contact the sender immediately and delete all copies of this transmission.

2 1

[Kamailio-Users] prepaid system
by igor 08 Jan '09

08 Jan '09

I need implement a prepaid system in my kamilio ! anybody help me !? any information ! Atenciosamente, Igor Marques celular: 55 21 7858 7499 nextel: 87*24074 e-mail: igor(a)carneiro.cc web: www.carneiro.cc -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

10 28

Re: [Kamailio-Users] how can I add .sar file as application to Wesip application server
by skumar453＠gmail.com 08 Jan '09

08 Jan '09

Hi thanku for reply while i am running the click2dial application i am getting the message like bellow ConvergedHostDeployer [main] - Installing web application at context path /click2call-08_11_06 from URL file:/usr/wesip/wesipapps/click2call-08_11_06 the log file geting while running the application is StandardAppSessionManager [main] - Force random number initialization completed processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:5075/ > Manager Wesip Application ==> http://localhost:5075/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml here i am not clear how to test this application ..can u please suggest me about this On Jan 7, 2009 8:27pm, Ginés Gómez <gines(a)voztele.com> wrote: > Hi, > > > > > > you can simply copy the .sar file at the wesipapps directory > > > > > > Regards > > > > > > Gines > > > > > > > > hi all > > > > > > I am workin on the "WeSIP" application server. I installed opensips as proxy with SEAS module and wesip as Application server. > > > these two are successfully connected ...but here i have no clue on how to add application "click2dial" to WeSIP. > > > > > > 1. please let me know how can I add .sar file as application. > > > 2. http://localhost:5075/manager/html? is showig a blank page. how can I get this page displayed? > > > > > > regards > > > suresh_______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > >

1 0

[Kamailio-Users] carrierroute prioritize gateways
by Juan Asencio 07 Jan '09

07 Jan '09

Hi list, I'm reading some old post in order to implement carrierroute. I found this in a post that I include at the end of this message, but the part that I want to ask about is this one: "You can put all the carriers under the same table and prioritize the gateway based on domains. Like tis you don't need to hardcode anything in the config file" If I do this, do I need to prioritize the gateways on the subscriber table and if is there, how do I do it for the domain, when it is only a cr_preferred_carrier field? Cheers, Juan.- [OpenSER-Users] Carrierroute *Ovidiu Sas* osas at voipembedded.com <mailto:users%40lists.kamailio.org?Subject=%5BOpenSER-Users%5D%20Carrierrout…> /Wed Mar 26 04:57:54 CET 2008/ * Previous message: [OpenSER-Users] Carrierroute <http://lists.opensips.org/pipermail/users/2008-March/016657.html> * Next message: [OpenSER-Users] [FEATURE REQ] allow does_uri_exist() to test against other variables besides $ru <http://lists.opensips.org/pipermail/users/2008-March/016658.html> * *Messages sorted by:* [ date ] <http://lists.opensips.org/pipermail/users/2008-March/date.html#16659> [ thread ] <http://lists.opensips.org/pipermail/users/2008-March/thread.html#16659> [ subject ] <http://lists.opensips.org/pipermail/users/2008-March/subject.html#16659> [ author ] <http://lists.opensips.org/pipermail/users/2008-March/author.html#16659> ------------------------------------------------------------------------ Hello Douglas, I am using this module in production with more then 400k routes loaded for 8 different providers and it is working like a charm. As Henning mentioned, the interface is a little bit cumbersome but usable. You can put all the carriers under the same table and prioritize the gateway based on domains. Like tis you don't need to hardcode anything in the config file. Carrierroute was designed to deal with a big amount of routing rules and it is doing an amazing job. LCR was designed to deal with a limited number of routes/gateways but it has a different level of flexibility and a more mature interface. Carrierroute is a brand new module and for sure it will improve in flexibility as times go by (just like the lcr did along releases). Hope this helps, Ovidiu Sas

2 3

[Kamailio-Users] dialplan and carrierroute
by victorpy 07 Jan '09

07 Jan '09

hello, sorry for the other message. i want to know if there is a way of routing based on patern matching using caller-id. Basically, i want to route through a carrier if the call-id have some pattern Eg.: international is 002-(country code)-(area code)-697699. Long distance is (country code) -(area code) -123456 Local is 123456 The international and long distance change If international then route through carrier1 else if long distance the route through carrier 2 else if local then route through carrier 3 -- View this message in context: http://www.nabble.com/dialplan-and-carrierroute-tp21331266p21331266.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Kamailio-Users] how can I add .sar file as application to Wesip application server
by skumar453＠gmail.com 07 Jan '09

07 Jan '09

hi all I am workin on the "WeSIP" application server. I installed opensips as proxy with SEAS module and wesip as Application server. these two are successfully connected ...but here i have no clue on how to add application "click2dial" to WeSIP. 1. please let me know how can I add .sar file as application. 2. http://localhost:5075/manager/html? is showig a blank page. how can I get this page displayed? regards suresh

2 1

[Kamailio-Users] moving my kamilio1.2 server to opensips
by Asim Riaz 07 Jan '09

07 Jan '09

Hi, could anyone advice me, I like to move my Kamalio 1.2 server to Opensips ? is it possible to move all the exisitng configuration transparently? Thanks in Advance. -Asim

3 3

[Kamailio-Users] Regarding 600 bussy every where when i Forward INVITe message from b2bua to scscf
by sureshkumar.jaiswal 07 Jan '09

07 Jan '09

Hi when i ran b2bua_simple.py like this python b2bua_simple.py -f -l 172.18.2.187 -p 99999 -n 172.18.2.187 b2bua_simple.py is running 172.18.2.187 and port for this is 99999 and scscf is also running 172.18.2.187:6060 and forward my INVITE message from scscf to b2bua_simple then b2bua got this INVITE message but when b2bua again forward to scscf then we getting 600 bussy every where and call has been terminated. folllowing message i have attache along this mail. can any one let me know what i have to do for successfull call using b2bua_simple.py. python b2bua_simple.py -f -l 172.18.2.187 -p 99999 -n 172.18.2.187 08 Jan 15:26:22/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:6060: INVITE sip:alice@info-spectrum.com SIP/2.0 Record-Route: <sip:mo@scscf.info-spectrum.com:6060;lr> Route: <sip:172.18.2.187:99999;lr>, <sip:iscmark@scscf.info-spectrum.com:6060;lr;s=1;h=0;d=0;a=7369703a626f6240696e666f2d737065637472756d2e636f6d> Record-Route: <sip:mo@pcscf.info-spectrum.com:4060;lr> Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 Via: SIP/2.0/UDP 172.18.2.187:4060;branch=z9hG4bK2ede.baaf9433.0 Via: SIP/2.0/UDP 172.18.2.74:2423;rport=2423;branch=z9hG4bK32209 From: <sip:bob@info-spectrum.com>;tag=9791 To: <sip:alice@info-spectrum.com> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 201 INVITE Max-Forwards: 15 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE,OPTIONS,UPDATE,NOTIFY,PRACK Contact: <sip:bob@172.18.2.74:2423;transport=udp>;expires=1000 User-Agent: Mercuro IMS Client Beta (4.0.1011.0) Supported: sdp-anat Content-Type: application/sdp P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=00000000 Privacy: none Content-Length: 468 P-Asserted-Identity: <sip:bob@info-spectrum.com> P-Charging-Vector: icid-value="P-CSCFabcd4965cdc6000000ea";icid-generated-at=172.18.2.187;orig-ioi="info-spectrum.com" Timestamp: 1231408582 v=0 o=- 3344 3344 IN IP4 172.18.2.74 s=Mercuro IMS Client Session t=0 0 m=audio 30904 RTP/AVP 0 8 101 c=IN IP4 172.18.2.74 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 30080 RTP/AVP 111 34 31 c=IN IP4 172.18.2.74 a=rtpmap:111 H263-1998/90000 a=fmtp:111 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=2;QCIF=2/MaxBR=2400 08 Jan 15:26:22/GLOBAL/b2bua: SENDING message to 172.18.2.187:6060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 Via: SIP/2.0/UDP 172.18.2.187:4060;branch=z9hG4bK2ede.baaf9433.0 Via: SIP/2.0/UDP 172.18.2.74:2423;rport=2423;branch=z9hG4bK32209 Record-Route: <sip:mo@scscf.info-spectrum.com:6060;lr> Record-Route: <sip:mo@pcscf.info-spectrum.com:4060;lr> From: <sip:bob@info-spectrum.com>;tag=9791 To: <sip:alice@info-spectrum.com> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 201 INVITE Server: Sippy B2BUA (Simple) 08 Jan 15:26:22/GLOBAL/b2bua: SENDING message to 172.18.2.187:5060: INVITE sip:alice@172.18.2.187:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.2.187:99999;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18;rport Max-Forwards: 70 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 INVITE Contact: Anonymous <sip:bob@172.18.2.187:99999> Expires: 300 User-Agent: Sippy B2BUA (Simple) cisco-GUID: 288073477-2250398923-1693218158-3417804826 h323-conf-id: 288073477-2250398923-1693218158-3417804826 Content-Length: 468 Content-Type: application/sdp v=0 o=- 3344 3344 IN IP4 172.18.2.74 s=Mercuro IMS Client Session t=0 0 m=audio 30904 RTP/AVP 0 8 101 c=IN IP4 172.18.2.74 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 30080 RTP/AVP 111 34 31 c=IN IP4 172.18.2.74 a=rtpmap:111 H263-1998/90000 a=fmtp:111 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=2;QCIF=2/MaxBR=2400 08 Jan 15:26:23/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.18.2.187:99999;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18;rport=34463 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 INVITE Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux)) Content-Length: 0 Timestamp: 1231408582 Warning: 392 172.18.2.187:5060 "Noisy feedback tells: pid=3661 req_src_ip=172.18.2.187 req_src_port=34463 in_uri=sip:alice@172.18.2.187:5060 out_uri=sip:alice@172.18.2.187:5060 via_cnt==1" 08 Jan 15:26:33/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:5060: SIP/2.0 600 Busy everywhere - Forwarding to S-CSCF failed Via: SIP/2.0/UDP 172.18.2.187:99999;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18;rport=34463 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187>;tag=0fb5d9176cb7d6a1cb577893687b8ec9-47d1 Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 INVITE Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux)) Content-Length: 0 Timestamp: 1231408593 Warning: 392 172.18.2.187:5060 "Noisy feedback tells: pid=3664 req_src_ip=172.18.2.187 req_src_port=5060 in_uri=sip:alice@172.18.2.187:5060 out_uri=sip:alice@172.18.2.187:5060 via_cnt==0" 08 Jan 15:26:33/GLOBAL/b2bua: SENDING message to 172.18.2.187:5060: ACK sip:alice@172.18.2.187:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.2.187:99999;rport;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18 Max-Forwards: 70 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187>;tag=0fb5d9176cb7d6a1cb577893687b8ec9-47d1 Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 ACK Expires: 300 User-Agent: Sippy B2BUA (Simple) 08 Jan 15:26:33/GLOBAL/b2bua: SENDING message to 172.18.2.187:6060: SIP/2.0 600 Busy everywhere - Forwarding to S-CSCF failed Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 Via: SIP/2.0/UDP 172.18.2.187:4060;branch=z9hG4bK2ede.baaf9433.0 Via: SIP/2.0/UDP 172.18.2.74:2423;rport=2423;branch=z9hG4bK32209 Record-Route: <sip:mo@scscf.info-spectrum.com:6060;lr> Record-Route: <sip:mo@pcscf.info-spectrum.com:4060;lr> From: <sip:bob@info-spectrum.com>;tag=9791 To: <sip:alice@info-spectrum.com>;tag=2ac46b4f62d11be3189628535aa7b381 Call-ID: 1231260517025(a)172.18.2.74 CSeq: 201 INVITE Server: Sippy B2BUA (Simple) 08 Jan 15:26:33/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:6060: ACK sip:alice@info-spectrum.com SIP/2.0 Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 From: <sip:bob@info-spectrum.com>;tag=9791 Call-ID: 1231260517025(a)172.18.2.74 To: <sip:alice@info-spectrum.com>;tag=2ac46b4f62d11be3189628535aa7b381 CSeq: 201 ACK Route: <sip:orig@scscf.info-spectrum.com:6060;lr> User-Agent: Sip EXpress router(2.1.0-dev1 OpenIMSCore (i386/linux)) Content-Length: 0

2 1

[Serusers] Large MTU Size of SIP Invites
by Jai Rangi 06 Jan '09

06 Jan '09

All, Greeting, I am using ser as a proxy to forward calls to Asterisk servers. When I forward the calls the total packet size is anywhere between 1505 to 1520. So the packets are frangmented. Some asterisks behind the firewall are not able to handle the franented sip packets of size 1500+. Is there a way to reduce the packets size to less than 1500. Any hint will be be appreciated. Thank you, Regards, -jai

2 1

[Kamailio-Users] dialplan and carrierroute
by victorpy 05 Jan '09

05 Jan '09

-- View this message in context: http://www.nabble.com/dialplan-and-carrierroute-tp21299057p21299057.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] extending carrierroute to load dynamic carriers
by victorpy 05 Jan '09

05 Jan '09

Hello, I want to extend the module to load a carrier tree based on $si(source ip) of a request. I want know if im going the right way doing this. The reason why im doing this is to avoid a hardcoded script. I have read other post that talk about this subject and the solution with domains and host doesn't meet my needs. Hope someone could help me. thanks for the answers greetings -- View this message in context: http://www.nabble.com/extending-carrierroute-to-load-dynamic-carriers-tp212… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 5

[Kamailio-Users] Setting answer timeout
by mayamatakeshi 05 Jan '09

05 Jan '09

Hello, I know I can customize the timeout for answer using fr_inv_timer: modparam("tm", "fr_inv_timer", 20) But is it possible to set the answer timeout on a transaction basis? (so that it can be customized by subscriber for example). regards, mayama

2 2

[Kamailio-Users] Owner of rtpproxy.sock
by Franz Edler 05 Jan '09

05 Jan '09

Hello experts! I have a problem with the owner of rtpproxy.sock. When I use the -u parameter at start of rtpproxy like "rtpproxy -s unix:/var/run/rtpproxy.sock -u sip:sip" the owner and group of the socket is still "root root". Is this correct? Should not it be sip sip? I want to avoid kamailio running as root and therefore I have to modify the permissions of rtpproxy.sock after start of rtpproxy accordingly. My question now is: Maybe I have a misunderstanding in the purpose of the -u parameter. What is the behaviour of the -u parameter? Can anyone shed some light on this? Regards Franz

2 4

[Kamailio-Users] Bring India together
by Sunkara RaviPrakash 04 Jan '09

04 Jan '09

1 0

[Kamailio-Users] Bring India together
by Sunkara RaviPrakash 04 Jan '09

04 Jan '09

1 0

[Kamailio-Users] trying to use call park for similar a music on hold
by Ricky Gutierrez 31 Dec '08

31 Dec '08

Hi list, I am trying to make work to the parking of calls with openser and asterisk, which I want to do is when two UAC are in the middle of a call one of them can transfer a call with *700 and this is sent to asterisk to the extension by default... the problem here is that when I make the transfer to the extension 700 the asterisk it doesn't return it to the extension that I originate the transfer, the call it returns to the extension in delay .. I can see when I make the transfer in the SDP that the openser puts me c=IN IP4 0.0.0.0 , but asterisk doesn't return the call to the extension that I originate the transfer U +2.857758 192.168.10.40:5060 -> 192.168.10.1:5060 INVITE sip:*700@192.168.10.1:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 To: <sip:*700@192.168.10.1>;tag=as5aea7a9e Contact: <sip:112@192.168.10.40:5060;transport=udp> Supported: replaces, timer, path Referred-By: <sip:120@192.168.10.38:5060> Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:*700@192.168.10.1:5070", nonce="495ac4ad695509e755aba895780497e8116e6353", response="40021b3138cbfefcd079505a55c6043f" Call-ID: f69f46f8461d45e0(a)192.168.10.40 CSeq: 55344 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 352 v=0 o=112 8001 8002 IN IP4 192.168.10.40 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 5006 RTP/AVP 0 18 3 97 2 9 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000646 192.168.10.1:5060 -> 192.168.10.40:5060 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 To: <sip:*700@192.168.10.1>;tag=as5aea7a9e Call-ID: f69f46f8461d45e0(a)192.168.10.40 CSeq: 55344 INVITE Server: OpenSER (1.3.4-notls (i386/linux)) Content-Length: 0 # U +0.000078 192.168.10.1:5060 -> 192.168.10.1:5070 INVITE sip:*700@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK537.cbf8a716.0 Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 To: <sip:*700@192.168.10.1>;tag=as5aea7a9e Contact: <sip:112@192.168.10.40:5060;transport=udp> Supported: replaces, timer, path Referred-By: <sip:120@192.168.10.38:5060> Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:*700@192.168.10.1:5070", nonce="495ac4ad695509e755aba895780497e8116e6353", response="40021b3138cbfefcd079505a55c6043f" Call-ID: f69f46f8461d45e0(a)192.168.10.40 CSeq: 55344 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 352 v=0 o=112 8001 8002 IN IP4 192.168.10.40 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 5006 RTP/AVP 0 18 3 97 2 9 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000263 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK537.cbf8a716.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 Record-Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 To: <sip:*700@192.168.10.1>;tag=as5aea7a9e Call-ID: f69f46f8461d45e0(a)192.168.10.40 CSeq: 55344 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*700@192.168.10.1:5070> Content-Length: 0 # U +0.000104 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK537.cbf8a716.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 Record-Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 To: <sip:*700@192.168.10.1>;tag=as5aea7a9e Call-ID: f69f46f8461d45e0(a)192.168.10.40 CSeq: 55344 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*700@192.168.10.1:5070> Content-Type: application/sdp Content-Length: 285 v=0 o=root 9758 9759 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 15948 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly # U +0.000104 192.168.10.1:5060 -> 192.168.10.40:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.40:5060;branch=z9hG4bK1a7bc3cdf6e22598 Record-Route: <sip:192.168.10.1;lr=on;ftag=0d380f28344df9f2> From: "Ventas" <sip:112@192.168.10.1>;tag=0d380f28344df9f2 To: <sip:*700@192.168.10.1>;tag=as5aea7a9e Call-ID: f69f46f8461d45e0(a)192.168.10.40 CSeq: 55344 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*700@192.168.10.1:5070> Content-Type: application/sdp Content-Length: 285 v=0 o=root 9758 9759 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 15948 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly # U +0.054311 192.168.10.40:5060 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK745a.fb2f15e.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK61abb915;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as6fb8efad> From: "Ventas" <sip:112@192.168.10.1>;tag=as6fb8efad To: <sip:112@192.168.10.1>;tag=368cbb40ae863e2a Call-ID: 3492c4a24c10596e6e3063c361c67eb9(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:112@192.168.10.40:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 234 v=0 o=112 8000 8000 IN IP4 192.168.10.40 s=SIP Call c=IN IP4 192.168.10.40 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 0 RTP/AVP 99 # U +0.000097 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK61abb915;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as6fb8efad> From: "Ventas" <sip:112@192.168.10.1>;tag=as6fb8efad To: <sip:112@192.168.10.1>;tag=368cbb40ae863e2a Call-ID: 3492c4a24c10596e6e3063c361c67eb9(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:112@192.168.10.40:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 234 I have openser and asterisk with realtime any ideas ? regards list rickygm

2 1

[Serusers] from header as append_hf() param
by caio 30 Dec '08

30 Dec '08

Hi all, Is it possible to add the current from header (%fu or $fu) or contact header as parameter of the append_hf textops function? I need to add a custom header like the following: append_hf("P-Asserted-Identity: \"Caller guy\" <%fu>); Result: ...P-Asserted-Identity: "Caller guy" <123(a)10.3.23.5>) There is an "append_urihf()" but it adds as prefix or suffix the R-URI address. Important, SER version 0.9.7. Regards, Claudio

1 0

[Serusers] get uri in module
by Netadmin Bensoft 30 Dec '08

30 Dec '08

Hello, I made a module and i want to get the uri from sip msg. How can i do this? What i need is the phone from uri or the uri (sip:1234@sip.sip.com) I've tried get_to(msg)->uri.s but it doesn't return the uri.. Thanks.

2 1

[Kamailio-Users] drop() and tm timers.
by Aurelien Grimaud 29 Dec '08

29 Dec '08

Hi, In my openser, I had to not relay any provisional which is < 180. I used drop in reply route to do this. No provisional < 180 is sent to caller. Unfortunately, dropping 100 seems to make the transaction module not disarming its fr_timer. The 180 has been waited for more time than my fr_timer so request failed and failure_route was called. In fact, there is no 180, only a 200 ok. Can anyone confirm that disarming timer is *not* done if 100 is dropped ? This is openser 1.2.1 Regards. Aurelien

3 10

[Kamailio-Users] "Failed" acc log when 183 Session Progress
by Samuel Muller 29 Dec '08

29 Dec '08

Hello all, I just have a little question about this thing : when I start a call, my Radius receives before the session is established a "failed" in its accounting log. If I move the setflag() for rad acc and rad failed acc (or one of each, then the other one), the "Start" disappear - but I get the Stop. How to not take acc in charge when there is a 183 Session Progress ? here is the radius acc log : Code: Accounting-Request Identifier: 102 Authentic: Z3z<16>^<230><251>i<233>[6Q<196><12><140><25> Attributes: Acct-Status-Type = Failed Service-Type = Sip-Session Sip-Response-Code = 183 Sip-Method = "<1>" Event-Timestamp = 1230128211 Sip-From-Tag = "a3addce43738cc27o0" Sip-To-Tag = "1c782477072" Acct-Session-Id = "10bdf568-2821ed73(a)192.168.1.7" Sip-Method = "INVITE" SER-From = "0170645745" SER-To = "0663128505" Sip-User-Digest-URI = "sip:0663128505@sip.720.fr<sip%3A0663128505(a)sip.720.fr> " Sip-Req-URI = "sip:0663128505@77.246.81.136:5060;transport=udp" Sip-Source-IP-Address = "87.90.4.63" Sip-Source-Port = "5060" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.60.9 the setflag() are here : if (is_method("INVITE")) { setflag(4); # flag "radius acc" setflag(5); # flag "radius failed acc" setflag(8); # flag "dialog" setflag(9); # flag "session timer" route(13); } the only place I found the related "183" taken in charge in the kamailio.cfg is here : onreply_route[2] { if (nat_uac_test("1")) { fix_nated_contact(); } if (isbflagset(14) && status =~ "(180|183|2[0-9][0-9])") #? branch flag nat { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } in the kamailio log for this same call : Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_msg: SIP Reply (status): Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_msg: version: <SIP/2.0> Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_msg: status: <100> Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_msg: reason: <Trying> Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: flags=2 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKbe63.1eddc1f3.0>; state=16 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_via: end of header reached, state=5 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: via found, flags=2 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: this is the first via Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:receive_msg: After parse_msg... Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:grep_sock_info: checking if host==us: 13==13 && [77.246.81.132] == [77.246.81.132] Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:forward_reply: found module tm, passing reply to it Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:t_check: start=0xffffffffffffffff Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: flags=22 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_via_param: found param type 235, <rport> = <5060>; state=6 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_via_param: found param type 234, <received> = <87.90.4.63>; state=6 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-8aeb8bb>; state=16 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_via: end of header reached, state=5 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: via found, flags=22 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: parse_headers: this is the second via Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_to_param: tag=1c782477072 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_to: end of header reached, state=29 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_to: display={"0663128505"}, ruri={sip:0663128505@sip.720.fr<sip%3A0663128505(a)sip.720.fr>} Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:get_hdr_field: <To> [58]; uri=[sip:0663128505@sip.720.fr <sip%3A0663128505(a)sip.720.fr>] Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:get_hdr_field: to body ["0663128505" <sip:0663128505@sip.720.fr <sip%3A0663128505(a)sip.720.fr>>] Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:get_hdr_field: cseq <CSeq>: <102> <INVITE> Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: flags=8 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:t_reply_matching: hash 14059 label 1058856417 branch 0 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:t_reply_matching: REF_UNSAFE: after is 1 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:t_reply_matching: reply matched (T=0x7fcc45ebb090)! Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:run_trans_callbacks: trans=0x7fcc45ebb090, callback type 2, id 1 entered Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:t_check: end=0x7fcc45ebb090 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=1) Dec 24 15:16:51 ser0 kamailio[7748]: -NAT-ONREPLY----------------------------- Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:parse_headers: flags=80 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:get_hdr_field: content_length=0 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:get_hdr_field: found end of header Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:t_should_relay_response: T_code=0, new_code=100 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:relay_reply: branch=0, save=0, relay=-1 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:set_timer: relative timeout is 60 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:insert_timer_unsafe: [1]: 0x7fcc45ebb2e0 (95) Dec 24 15:16:51 ser0 kamailio[7748]: DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:destroy_avp_list: destroying list (nil) Dec 24 15:16:51 ser0 kamailio[7748]: DBG:core:receive_msg: cleaning up Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_msg: SIP Reply (status): Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_msg: version: <SIP/2.0> Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_msg: status: <183> Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_msg: reason: <Session Progress> Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=2 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKbe63.1eddc1f3.0>; state=16 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_via: end of header reached, state=5 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: via found, flags=2 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: this is the first via Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:receive_msg: After parse_msg... Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:grep_sock_info: checking if host==us: 13==13 && [77.246.81.132] == [77.246.81.132] Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:forward_reply: found module tm, passing reply to it Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:t_check: start=0xffffffffffffffff Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=22 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_via_param: found param type 235, <rport> = <5060>; state=6 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_via_param: found param type 234, <received> = <87.90.4.63>; state=6 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-8aeb8bb>; state=16 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_via: end of header reached, state=5 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: via found, flags=22 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: parse_headers: this is the second via Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_to_param: tag=1c782477072 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_to: end of header reached, state=29 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_to: display={"0663128505"}, ruri={sip:0663128505@sip.720.fr<sip%3A0663128505(a)sip.720.fr>} Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:get_hdr_field: <To> [58]; uri=[sip:0663128505@sip.720.fr <sip%3A0663128505(a)sip.720.fr>] Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:get_hdr_field: to body ["0663128505" <sip:0663128505@sip.720.fr <sip%3A0663128505(a)sip.720.fr>>] Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:get_hdr_field: cseq <CSeq>: <102> <INVITE> Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=8 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:t_reply_matching: hash 14059 label 1058856417 branch 0 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:t_reply_matching: REF_UNSAFE: after is 1 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:t_reply_matching: reply matched (T=0x7fcc45ebb090)! Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:run_trans_callbacks: trans=0x7fcc45ebb090, callback type 2, id 1 entered Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=8 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:t_check: end=0x7fcc45ebb090 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=0 is_invite=1) Dec 24 15:16:51 ser0 kamailio[7747]: -NAT-ONREPLY----------------------------- Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=80 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=ffffffffffffffff Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:get_hdr_field: content_length=293 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:get_hdr_field: found end of header Dec 24 15:16:51 ser0 kamailio[7747]: DBG:nathelper:check_content_type: type <application/sdp> found valid Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=40 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_to_param: tag=a3addce43738cc27o0 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_to: end of header reached, state=29 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_to: display={"0170645745"}, ruri={sip:0170645745@sip.720.fr<sip%3A0170645745(a)sip.720.fr>} Dec 24 15:16:51 ser0 kamailio[7747]: DBG:nathelper:force_rtp_proxy2_f: proxy reply: 35934 77.246.81.133 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:t_should_relay_response: T_code=0, new_code=183 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:relay_reply: branch=0, save=0, relay=0 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:run_trans_callbacks: trans=0x7fcc45ebb090, callback type 16, id 0 entered Dec 24 15:16:51 ser0 kamailio[7747]: DBG:dialog:run_dlg_callbacks: dialog=0x7fcc45ebc7c0, type=256 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:sst:sst_dialog_response_fwded_CB: Dialog seen REPLY 183 Session Progress Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=2000 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=ffffffffffffffff Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:clen_builder: content-length: 311 (311) Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:build_res_buf_from_sip_res: old size: 1027, new size: 983 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:build_res_buf_from_sip_res: copied size: orig:1025, new: 981, rest: 2 msg= SIP/2.0 183 Session Progress^M Via: SIP/2.0/UDP 192.168.1.7:5060;rport=5060;received=87.90.4.63;branch=z9hG4bK-8aeb8bb^M From: "0170645745" <sip:0170645745@sip.720.fr <sip%3A0170645745(a)sip.720.fr>>;tag=a3addce43738cc27o0^M To: "0663128505" <sip:0663128505@sip.720.fr <sip%3A0663128505(a)sip.720.fr>>;tag=1c782477072^M Call-ID: 10bdf568-2821ed73(a)192.168.1.7^M CSeq: 102 INVITE^M Contact: < sip:61@77.246.81.136 <sip%3A61(a)77.246.81.136>>^M Record-Route: <sip:77.246.81.132;lr=on;ftag=a3addce43738cc27o0;nat=yes;did=17a.36b075d2>^M Supported: em,timer,replaces,path,resource-priority^M Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE^M Server: 720 DEGRES PSTN GW/v.5.40A.030.002^M Content-Type: application/sdp^M Content-Length: 311^M ^M v=0^M o=AudiocodesGW 782585440 782585110 IN IP4 77.246.81.136^M s=Phone-Call^M c=IN IP4 77.246.81.133^M t=0 0^M m=audio 35934 RTP/AVP 18 101^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=yes^M a=rtpmap:101 telephone-event/ Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:_shm_resize: resize(0) called Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:relay_reply: sent buf=0x7936d8: SIP/2.0 1..., shmem=0x7fcc45ebef00: SIP/2.0 1 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:run_trans_callbacks: trans=0x7fcc45ebb090, callback type 128, id 1 entered Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:parse_headers: flags=14000 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:run_trans_callbacks: trans=0x7fcc45ebb090, callback type 128, id 0 entered Dec 24 15:16:51 ser0 kamailio[7747]: DBG:dialog:next_state_dlg: dialog 0x7fcc45ebc7c0 changed from state 1 to state 2, due event 2 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:set_timer: relative timeout is 60 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:insert_timer_unsafe: [1]: 0x7fcc45ebb2e0 (96) Dec 24 15:16:51 ser0 kamailio[7747]: DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:destroy_avp_list: destroying list (nil) Dec 24 15:16:51 ser0 kamailio[7747]: DBG:core:receive_msg: cleaning up Dec 24 15:16:51 ser0 kamailio[7753]: DBG:tm:utimer_routine: timer routine:4,tl=0x7fcc45ebb2b0 next=(nil), timeout=36400000 -- Samuel MULLER sml(a)720.fr

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[Kamailio-Users] snmpstats error
by Stagg Shelton 28 Dec '08

28 Dec '08

I am planning an upgrade from openser 1.3.4 to kamailio 1.4.3, and I am having an issue starting kamailio with snmpstats enabled. I find the errors below in the kamailio syslog file. It looks like the root issue is a missing symbol, but I have not been able to figure out what provides the symbol. Openser 1.3.4 works fine on the same machine with essentially the same configuration that I am wanting to use for kamailio. The snmpstats.so module compiled without any errors, and is located in the /lib/ kamailio/modules/ directory. The system is fedora core 6 fully yum updated. Dec 24 14:24:26 sip-interop-c61-atl kamailio: ERROR:core:sr_load_module: could not open module <//lib/kamailio/ modules/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: make_tempfile Dec 24 14:24:26 sip-interop-c61-atl kamailio: CRITICAL:core:yyerror: parse error in config file, line 57, column 13-14: failed to load module Dec 24 14:24:26 sip-interop-c61-atl kamailio: ERROR:core:set_mod_param_regex: no module matching snmpstats found | Dec 24 14:24:26 sip-interop-c61-atl kamailio: CRITICAL:core:yyerror: parse error in config file, line 142, column 20-21: Can't set module parameter Dec 24 14:24:26 sip-interop-c61-atl kamailio: ERROR:core:set_mod_param_regex: no module matching snmpstats found | Dec 24 14:24:26 sip-interop-c61-atl kamailio: CRITICAL:core:yyerror: parse error in config file, line 143, column 20-21: Can't set module parameter Dec 24 14:24:26 sip-interop-c61-atl kamailio: ERROR:core:set_mod_param_regex: no module matching snmpstats found | Dec 24 14:24:26 sip-interop-c61-atl kamailio: CRITICAL:core:yyerror: parse error in config file, line 144, column 20-21: Can't set module parameter Dec 24 14:24:26 sip-interop-c61-atl kamailio: ERROR:core:set_mod_param_regex: no module matching snmpstats found | Dec 24 14:24:26 sip-interop-c61-atl kamailio: CRITICAL:core:yyerror: parse error in config file, line 145, column 20-21: Can't set module parameter Dec 24 14:24:26 sip-interop-c61-atl kamailio: ERROR:core:main: bad config file (5 errors) Has anyone seen this error, or something similar? Thank You Stagg Shelton www.sheltonjohns.com

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[Kamailio-Users] sub:Query on PUA
by skumar453＠gmail.com 28 Dec '08

28 Dec '08

Hi All, HAPPY CHRISTMAS , I am working on PUA module.I am running PUA as Apllication server.I have registered ALICE in a client which doesn't have presence support (OpenIC client) and BOB in UCTIMSCLIENT which has presence support.And i am invoking SUBSCRIBE message by a script file to know the presence of BOB to ALICE. When the status of BOB is changed,NOTIFY is getting Terminated at I-CSCF. ERROR i am getting is : "406 Initial Request Method not allowed atthe I-CSCF". According to traces while registration flow is as follows: FLOW 1: PUA------>i-cscf------>s-cscf. But actually the flow should be: FLOW 2: PUA------>p-cscf------>s-cscf. I have enabled outbound proxy server in configuration file to get FLOW 2.But,the flow is not changed. Kindly let me know ,how to get NOTIFY message when status of BOB is changed. Thanks in Advance.

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Re: [Kamailio-Users] [Kamailio-Devel] new module: qos
by Daniel-Constantin Mierla 28 Dec '08

28 Dec '08

Thanks Ovidiu, readme is online at: http://kamailio.org/docs/modules/devel/qos.html Is the module keeping track of all session negotiated, or is stores the last one (used)? Happy holidays! Daniel On 12/23/08 23:05, Ovidiu Sas wrote: > Hello, > > QoS is an new module that can be used to track per dialog SDP sessions. > The QoS module is sitting on top of the dialog module and provides an > API to be used by other module. > The QoS module does not provide any scripting functionality (it is a > passive module). > See the module documentation for more informations. > > SDP session(s) can be inspected via the dialog dlg_list_ctx MI > interface command: > dialog:: hash=3732:573385728 > state:: 4 > timestart:: 1230068558 > timeout:: 1009 > callid:: d3d816cce3e24a19b937b56e4689e524 > from_uri:: sip:111111111@10.11.10.63 > from_tag:: 8c85c079 > caller_contact:: sip:111111111@10.11.128.129 > caller_cseq:: 100 > caller_route_set:: > caller_bind_addr:: udp:10.11.10.63:5060 > to_uri:: sip:222222222@10.11.10.63 > to_tag:: 43FED9C4-20CF > callee_contact:: sip:222222222@10.111.3.241:5060 > callee_cseq:: 100 > callee_route_set:: > callee_bind_addr:: udp:10.11.10.63:5060 > context:: > qos:: negotiated_sdp > sdp:: m_dir=1 m_id=1 method=INVITE cseq=100 > negotiation=1 > session:: callee cnt_disp= bw_type= > bw_width= streams=2 > stream:: 1 media=video > IP=10.111.3.241 port=0 transport=RTP/AVP payloads_num=0 > stream:: 0 media=audio > IP=10.111.3.241 port=17240 transport=RTP/AVP payloads_num=3 > payload:: 2 rtpmap=100 > codec=X-NSE > payload:: 1 rtpmap=101 > codec=telephone-event > payload:: 0 rtpmap=0 codec=PCMU > session:: caller cnt_disp= bw_type=AS > bw_width=400 streams=2 > stream:: 1 media=video > IP=10.11.128.129 port=22804 transport=RTP/AVP payloads_num=2 > payload:: 1 rtpmap=34 codec=H263 > payload:: 0 rtpmap=107 > codec=H263-1998 > stream:: 0 media=audio > IP=10.11.128.129 port=22800 transport=RTP/AVP payloads_num=12 > payload:: 11 > rtpmap=112 codec=iLBC > payload:: 10 > rtpmap=100 codec=AMR > payload:: 9 rtpmap=13 codec=CN > payload:: 8 rtpmap=101 > codec=telephone-event > payload:: 7 rtpmap=111 > codec=G726-40 > payload:: 6 rtpmap=110 > codec=G726-32 > payload:: 5 rtpmap=109 > codec=G726-24 > payload:: 4 rtpmap=108 > codec=G726-16 > payload:: 3 rtpmap=4 codec=G723 > payload:: 2 rtpmap=18 codec=G729 > payload:: 1 rtpmap=0 codec=PCMU > payload:: 0 rtpmap=8 codec=PCMA > > > > Regards, > Ovidiu Sas > > _______________________________________________ > Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Merry Christmas and Happy New Year 2009!
by Daniel-Constantin Mierla 24 Dec '08

24 Dec '08

Hello, I wish everyone Merry Christmas and Happy New Year 2009! Day by day, we ended another year. 2008 was full of events, starting with one major release and six minor releases, participation to workshops and conferences wordlwide, adoption in new production environments and acknowledgement of running platforms with millions of users and billions of routed minutes/month using Kamailio (OpenSER), check the News archive at: http://www.kamailio.org There were sad events as well, including the hijacking of the old domain name after the rename in Kamailio, still the source forge project goes on, with same name, proving the genuine openser project: https://sourceforge.net/projects/openser/ We maintained release 1.3.x series, last being 1.3.4, which still keeps the old name and is available at: http://www.openser-project.org/mos/view/News/NewsItem/OpenSER-v1.3.4-Releas… Life is going on, since the latest major release, version 1.4.0, the project boosted in new features, a summary of what is new since then is available here: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x An important share of effort was directed to clean the code and improve the stability and performance, a proof of maturity and the need to make the maintenance of the project easier and open for new comers. Lot of documentation was added in doxygen format inside the source code, Devel Guide was published, new developers can start new easier to enhance the application. Reference moment for the future was the launch of SIP Router project in November - a concentration of many people with great SIP and VoIP expertise, backed up by huge programming experience in this field. This came to strengthen the development workforce, to ensure the reliability, to remove the doubts of what projects is better now or is going to develop better in the future. http://sip-router.org So, 2009 is an year that announces already great achievements to be done in its first half: - 1st Quarter - release of Kamailio (OpenSER) 1.5.0 - 2nd Quarter - release of SIP Router project as stable version - dedicated meeting to celebrate the first operational SIP Router and draw future directions With the release of SIP Router project, everyone gets access to the features provided by Kamailio (OpenSER), SIP Express Router and OpenIMSCore projects: http://sip-router.org/benefits/ Looking forward to a fruitful 2009! Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] how to bild "pidf" for network elements
by skumar453＠gmail.com 22 Dec '08

22 Dec '08

Hi all I am developing presence network agent(PNA)..IT collects the presence information from the core network elements like Serving-call session control function(scscf)..and home subscriber server(HSS). and it will construct the publish message and send it to the Presence server(PS). here my doubt is what should be the content in the publish message for network presence? can any one suggest how to do it ? and can anyone let me know, whether there are any RFC or Specification for constructing PUBLISH message for network elements? thank you in advance suresh

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[Kamailio-Users] Using a b2bua as a prepaid box
by Luciano Afranllie 22 Dec '08

22 Dec '08

Hi everybody, I am experimenting trying to build a prepaid solution using prepaid extension for RADIUS (IETF draft). My current test bed is Kamailio as SIP proxy/registrar, asterisk as PSTN gateway, sippy b2bua and a freeRadius server. What I am thinking about is to split users in groups (prepaid, postpaid and suspended, for example) and to send all calls from prepaid users to the b2bua. The b2bua will have the prepaid client logic to enforce balance usage and drop calls. I am new in this area so I will appreciate a lot your feedback about what is the best way to do this. - Regarding groups. Is it worth to use radius for this or can I just use kamailio groups? - Regarding sending prepaid users calls to b2bua. It is a nice solution in kamailio to check user group and if it belongs to prepaid group forward the call to b2bua and b2bua forward the call again to kamailio in order to make the final routing? How should I identify the call has already been sent to the b2bua and is coming again to kamailio, just using the source host, or there is a better way? Thanks for your time. L:uciano

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[Kamailio-Users] dns updates
by Daniel-Constantin Mierla 21 Dec '08

21 Dec '08

Hello, there will be some DNS updates to kamailio.org domain during this weekend, so there might be some outages until the DNS propagates. You can safely use kamailio.net to browse the web all the time. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Fw: Help!! Delay in Invite with Authentication
by Rosario Pingaro 21 Dec '08

21 Dec '08

i am helping krunal debug this issue. The log from the router seems fine, no dealy in any other kind of messages. As the packets leave the router after 15ms they reach the proxy. Logging the router we get that the new invite is sent with delay, Is it possible that the router has some problem preparing the digest parameters? The very strange think is that the same router with the same config changing only the default proxy and letting it to point to a old SER config that does the invita auth by the same way, we don't get any delay. regards Rosario Pingaro D. Lgs 196/2003 Il presente messaggio contiene informazioni confidenziali, indirizzate esclusivamente alle persone sopra indicate. Se il ricevente non è tra dette persone, non dovrà intraprendere alcuna azione, tipo copia, stampa o trasmettere il suo contenuto a terzi ed i relativi allegati, ma solo informare il mittente dell'errore e cancellare il messaggio. Il mittente dovrà, altresì, accertarsi che gli allegati non contengano virus prima di aprirli. > ----- Original Message ----- > From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> > To: "Krunal Patel" <krunal.lists(a)gmail.com> > Cc: <users(a)lists.kamailio.org> > Sent: Saturday, December 20, 2008 10:20 AM > Subject: Re: [Kamailio-Users] Help!! Delay in Invite with Authentication > > >> Hello, >> >> On 12/19/08 15:40, Krunal Patel wrote: >>> Hi, >>> >>> I am using openser 1.2.3. >>> >>> From a router calls comes to openser. >>> Router is PATTON 4960. >>> I am using domain name to send calls to openser. >>> >>> First invite comes to openser & openser sends 407 to the router & then >>> getting ACK from the router to openser. >>> >>> Now the invite with authentication comes after almost 10 to 11 sec >>> Since ACK comes to openser. >> looks being related to the client. Can you check the logs of the router? >> Might be something with the DNS, responding slow. >> >> Also, if you can plug the router in a hub you can check when the >> messages are sent and when they reach the sip proxy, being able to see >> where is the issue. >> >> Cheers, >> Daniel >> >>> >>> Here is the trace: >>> # >>> U 2008/12/19 12:24:24.374228 XXX.XXX.XXX.XXX:5060 -> >>> YYY.YYY.YYY.YYY:5060 >>> INVITE sip:[TO_NUMBER]@[EMAIL PROTECTED] SIP/2.0. >>> Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK227032cab. >>> Max-Forwards: 70. >>> Content-Length: 341. >>> To: sip:[TO_NUMBER]@[EMAIL PROTECTED]. >>> From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. >>> ... >>> # >>> U 2008/12/19 12:24:24.374544 YYY.YYY.YYY.YYY:5060 -> >>> XXX.XXX.XXX.XXX:5060 >>> SIP/2.0 407 Proxy Authentication Required. >>> Via: SIP/2.0/UDP >>> XXX.XXX.XXX.XXX:5060;branch=z9hG4bK227032cab;rport=5060. >>> To: sip:[TO_NUMBER]@[EMAIL >>> PROTECTED];tag=b802e73ab92fec121a4f880263d541a3.7575. >>> From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. >>> ... >>> >>> # >>> U 2008/12/19 12:24:24.407646 XXX.XXX.XXX.XXX:5060 -> >>> YYY.YYY.YYY.YYY:5060 >>> ACK sip:[TO_NUMBER]@[EMAIL PROTECTED] SIP/2.0. >>> Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK227032cab. >>> Max-Forwards: 70. >>> Content-Length: 0. >>> To: sip:[TO_NUMBER]@[EMAIL >>> PROTECTED];tag=b802e73ab92fec121a4f880263d541a3.7575. >>> Route: <sip:[EMAIL PROTECTED]:5060;lr>. >>> .... >>> >>> >>> # >>> U 2008/12/19 12:24:35.667448 XXX.XXX.XXX.XXX:5060 -> >>> YYY.YYY.YYY.YYY:5060 >>> INVITE sip:[TO_NUMBER]@[EMAIL PROTECTED] SIP/2.0. >>> Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK3362d177f. >>> Max-Forwards: 70. >>> Content-Length: 341. >>> To: sip:[TO_NUMBER]@[EMAIL PROTECTED]. >>> From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. >>> Route: <sip:[EMAIL PROTECTED]:5060;lr>. >>> ... >>> >>> U 2008/12/19 12:24:35.692576 YYY.YYY.YYY.YYY:5060 -> >>> ZZZ.ZZZ.ZZZ.ZZZ:5060 >>> INVITE sip:[TO_NUMBER]@ZZZ.ZZZ.ZZZ.ZZZ:5060;transport=udp SIP/2.0. >>> Record-Route: <sip:[TO_NUMBER]@[EMAIL >>> PROTECTED]:5060;ftag=72a0c76ae3d477c;lr=on>. >>> Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK32dd.6cc062c5.0. >>> Via: SIP/2.0/UDP >>> XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bK3362d177f. >>> Max-Forwards: 69. >>> Content-Length: 341. >>> To: sip:[TO_NUMBER]@[EMAIL PROTECTED]. >>> From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. >>> ... >>> >>> U 2008/12/19 12:24:35.731432 ZZZ.ZZZ.ZZZ.ZZZ:5060 -> >>> YYY.YYY.YYY.YYY:5060 >>> SIP/2.0 100 Trying. >>> Call-ID: 1c00f084ccd0232ad5ab638dc6a7ccfd@[EMAIL PROTECTED]. >>> CSeq: 1965370819 INVITE. >>> From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. >>> .... >>> >>> >>> U 2008/12/19 12:24:35.849714 ZZZ.ZZZ.ZZZ.ZZZ:5060 -> >>> YYY.YYY.YYY.YYY:5060 >>> SIP/2.0 183 Session Progress. >>> Call-ID: 1c00f084ccd0232ad5ab638dc6a7ccfd@[EMAIL PROTECTED]. >>> CSeq: 1965370819 INVITE. >>> From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. >>> To: sip:[TO_NUMBER]@[EMAIL PROTECTED];tag=86f125f633683d0. >>> ... >>> >>> # >>> U 2008/12/19 12:24:35.849961 YYY.YYY.YYY.YYY:5060 -> >>> XXX.XXX.XXX.XXX:5060 >>> SIP/2.0 183 Session Progress. >>> Call-ID: 1c00f084ccd0232ad5ab638dc6a7ccfd@[EMAIL PROTECTED]. >>> CSeq: 1965370819 INVITE. >>> From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. >>> To: sip:[TO_NUMBER]@[EMAIL PROTECTED];tag=86f125f633683d0. >>> >>> >>> Can anybody help me to trobleshoot the issue? >>> >>> Thanks in Advance!! >>> >>> -- >>> Krunal patel >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

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[Serusers] error in acc_radius in User-Name AVP - ser rel_2_0_0
by Tomasz Zieleniewski 20 Dec '08

20 Dec '08

Hi, There is an error in acc_radius module when SER sends first accounting request corresponding to invite the User-Name AVP is taken as the "user" parameter in the Authorization header field, which is the correct behaviour. Acct-Session-Id = "Njg1M2M4NzE0NzAyZTIwZjcyOWNhMzIyMGVlY2RiNmQ." Sip-To-Tag = "0234593C-494D655A00008C74-40C0B950" SER-From = "\"Tomasz Zieleniewski\"<sip:tzl@sip.rd.touk.pl<sip%3Atzl(a)sip.rd.touk.pl> >;tag=815fbd6e" SER-Flags = 140509184 SER-Original-Request-ID = "sip:test@sip.rd.touk.pl<sip%3Atest(a)sip.rd.touk.pl> " Sip-Method = "INVITE" Sip-Cseq = 2 Sip-Translated-Request-ID = "sip:announcement@212.180.179.42:7160" Sip-Source-IP-Address = 192.168.1.132 Sip-From-Tag = "815fbd6e" SER-To = "\"test\"<sip:test@sip.rd.touk.pl<sip%3Atest(a)sip.rd.touk.pl> >;tag=0234593C-494D655A00008C74-40C0B950" SER-Digest-Username = "tzl" SER-Request-Timestamp = 1229808986 Calling-Station-Id = "sip:tzl@sip.rd.touk.pl<sip%3Atzl(a)sip.rd.touk.pl> " Sip-Source-Port = 5499 SER-Digest-Realm = "touk.pl" Sip-Response-Code = 200 Called-Station-Id = "sip:test@sip.rd.touk.pl<sip%3Atest(a)sip.rd.touk.pl> " SER-Response-Timestamp = 1229808986 Acct-Status-Type = Start Service-Type = IAPP-Register User-Name = "tzl(a)touk.pl" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 When there is another accounting request for instance when there is session termination by BYE request User-Name AVP is equal to the "From" Header when it shouldn't because it should be used if there was no authentication. Acct-Session-Id = "Njg1M2M4NzE0NzAyZTIwZjcyOWNhMzIyMGVlY2RiNmQ." Sip-To-Tag = "0234593C-494D655A00008C74-40C0B950" SER-From = "\"Tomasz Zieleniewski\"<sip:tzl@sip.rd.touk.pl<sip%3Atzl(a)sip.rd.touk.pl> >;tag=815fbd6e" SER-Flags = 140509184 SER-Original-Request-ID = "sip:announcement@212.180.179.42:7160" Sip-Method = "BYE" Sip-Cseq = 3 Sip-Translated-Request-ID = "sip:announcement@212.180.179.42:7160" Sip-Source-IP-Address = 192.168.1.132 Sip-From-Tag = "815fbd6e" SER-To = "\"test\"<sip:test@sip.rd.touk.pl<sip%3Atest(a)sip.rd.touk.pl> >;tag=0234593C-494D655A00008C74-40C0B950" SER-Request-Timestamp = 1229808992 Calling-Station-Id = "sip:tzl@sip.rd.touk.pl<sip%3Atzl(a)sip.rd.touk.pl> " Sip-Source-Port = 5499 Sip-Response-Code = 200 Called-Station-Id = "sip:test@sip.rd.touk.pl<sip%3Atest(a)sip.rd.touk.pl> " SER-Response-Timestamp = 1229808992 Acct-Status-Type = Stop Service-Type = IAPP-Register User-Name = "tzl(a)sip.rd.touk.pl" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Kind regards Tomasz Zieleniewski

1 0

[Kamailio-Users] Kamailio 1.4.2 : Weird Retransmission
by Aurelien Grimaud 20 Dec '08

20 Dec '08

Hi, I am using kamailio 1.4.2 and testing it with sipp. one sipp is client a uac (port 5066), the other is a uas (port 5063). kamailio is using 5060. I am generating 100 calls per second for 1000 calls and message were retransmitted 4 times. Here is a tcpdump of sip packets. Can anyone tell me why there is an INVITE retransmission at 16:16:38.627198 though the ACK for invite was received at 16:16:38.126096 ? Is there a known problem with retransmission timers ? 16:16:38.120384 IP 192.168.168.33.5066 > 192.168.168.33.5060: SIP, length: 579 E.._..@.@.f....!...!.....K..INVITE sip:otest@testg5 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5066;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5> Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 70 Expires: 180 Subject: Performance Test Content-Type: application/sdp Content-Length: 188 v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 8925 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 16:16:38.123426 IP 192.168.168.33.5060 > 192.168.168.33.5066: SIP, length: 325 E..a..@.@.g....!...!.....M..SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 127.0.0.1:5066;branch=z9hG4bK-23096-730-1;received=192.168.168.33 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5> Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Server: Kamailio (1.4.2-notls (i386/linux)) Content-Length: 0 16:16:38.123554 IP 192.168.168.33.5060 > 192.168.168.33.5063: SIP, length: 731 E.....@.@.fZ...!...!........INVITE sip:otest@testg5 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5> Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Expires: 180 Subject: Performance Test Content-Type: application/sdp Content-Length: 188 v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 8925 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 16:16:38.123830 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 451 E.....@.@.gz...!...!.......pSIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.0, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:38.124169 IP 192.168.168.33.5060 > 192.168.168.33.5066: SIP, length: 393 E.....@.@.g....!...!.......6SIP/2.0 180 Ringing Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:38.124326 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 634 E.....@.@.f....!...!.......'SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.0, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:127.0.0.1:5063> Content-Type: application/sdp Content-Length: 153 v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 6004 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 16:16:38.124883 IP 192.168.168.33.5060 > 192.168.168.33.5066: SIP, length: 581 E..a..@.@.f....!...!.....M..SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:192.168.168.33:5063> Content-Type: application/sdp Content-Length: 153 v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 6004 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 16:16:38.125532 IP 192.168.168.33.5066 > 192.168.168.33.5060: SIP, length: 427 E.....@.@.g....!...!.......XACK sip:192.168.168.33:5063 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5066;branch=z9hG4bK-23096-730-11 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696 Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 16:16:38.126096 IP 192.168.168.33.5060 > 192.168.168.33.5063: SIP, length: 521 E..%..@.@.g,...!...!........ACK sip:192.168.168.33:5063 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.2 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-11 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Subject: Performance Test Content-Length: 0 16:16:38.627198 IP 192.168.168.33.5060 > 192.168.168.33.5063: SIP, length: 731 E.....@.@.fZ...!...!........INVITE sip:otest@testg5 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5> Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Expires: 180 Subject: Performance Test Content-Type: application/sdp Content-Length: 188 v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 8925 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 16:16:38.627677 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 474 E.....@.@.gc...!...!........SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.2, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-11 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:38.627970 IP 192.168.168.33.5060 > 192.168.168.33.5066: SIP, length: 416 E.....@.@.g....!...!.......MSIP/2.0 180 Ringing Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-11 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:39.642356 IP 192.168.168.33.5060 > 192.168.168.33.5063: SIP, length: 731 E.....@.@.fZ...!...!........INVITE sip:otest@testg5 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5> Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Expires: 180 Subject: Performance Test Content-Type: application/sdp Content-Length: 188 v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 8925 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 16:16:39.642587 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 474 E.....@.@.gc...!...!........SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.2, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-11 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:39.643723 IP 192.168.168.33.5060 > 192.168.168.33.5066: SIP, length: 416 E.....@.@.g....!...!.......MSIP/2.0 180 Ringing Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-11 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5>;tag=17278SIPpTag01696;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:39.797641 IP 192.168.168.33.5066 > 192.168.168.33.5060: SIP, length: 434 E.....@.@.g....!...!......._BYE sip:127.0.0.1:5063 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5066;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 16:16:39.798436 IP 192.168.168.33.5060 > 127.0.0.1.5063: SIP, length: 528 E..,..@.@.P....!............BYE sip:127.0.0.1:5063 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Subject: Performance Test Content-Length: 0 16:16:39.798535 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 394 E.....@.@.g....!...!.......7SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:39.798814 IP 192.168.168.33.5060 > 192.168.168.33.5066: SIP, length: 336 E..l..@.@.g....!...!.....X..SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:40.257973 IP 192.168.168.33.5060 > 127.0.0.1.5063: SIP, length: 528 E..,..@.@.P....!............BYE sip:127.0.0.1:5063 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Subject: Performance Test Content-Length: 0 16:16:40.258080 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 394 E.....@.@.g....!...!.......7SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:41.285493 IP 192.168.168.33.5060 > 127.0.0.1.5063: SIP, length: 528 E..,..@.@.P....!............BYE sip:127.0.0.1:5063 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Subject: Performance Test Content-Length: 0 16:16:41.285593 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 394 E.....@.@.g....!...!.......7SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: <sip:127.0.0.1:5063> Content-Length: 0 16:16:41.695686 IP 192.168.168.33.5060 > 192.168.168.33.5063: SIP, length: 731 E.....@.@.fZ...!...!........INVITE sip:otest@testg5 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bKcb7d.7c08af34.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-1 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:otest@testg5> Call-ID: 730-23096-127.0.0.1 CSeq: 1 INVITE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Expires: 180 Subject: Performance Test Content-Type: application/sdp Content-Length: 188 v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 8925 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 16:16:43.338592 IP 192.168.168.33.5060 > 127.0.0.1.5063: SIP, length: 528 E..,..@.@.P....!............BYE sip:127.0.0.1:5063 SIP/2.0 Record-Route: <sip:192.168.168.33;lr=on;ftag=23096SIPpTag09730> Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0 Via: SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: sip:sipp@127.0.0.1:5066 Max-Forwards: 69 Subject: Performance Test Content-Length: 0 16:16:43.338785 IP 192.168.168.33.5063 > 192.168.168.33.5060: SIP, length: 408 E.....@.@.g....!...!.......ESIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.168.33;branch=z9hG4bK9b7d.0423bf25.0, SIP/2.0/UDP 127.0.0.1:5066;received=192.168.168.33;branch=z9hG4bK-23096-730-46 From: sipp <sip:730-23096-127.0.0.1@127.0.0.1:5066>;tag=23096SIPpTag09730 To: sut <sip:service@192.168.168.33:5060>;tag=17278SIPpTag01696 Call-ID: 730-23096-127.0.0.1 CSeq: 2 BYE Contact: <sip:127.0.0.1:5063;transport=UDP> Content-Length: 0

5 47

[Kamailio-Users] Help!! Delay in Invite with Authentication
by Krunal Patel 20 Dec '08

20 Dec '08

Hi, I am using openser 1.2.3. >From a router calls comes to openser. Router is PATTON 4960. I am using domain name to send calls to openser. First invite comes to openser & openser sends 407 to the router & then getting ACK from the router to openser. Now the invite with authentication comes after almost 10 to 11 sec Since ACK comes to openser. Here is the trace: # U 2008/12/19 12:24:24.374228 XXX.XXX.XXX.XXX:5060 -> YYY.YYY.YYY.YYY:5060 INVITE sip:[TO_NUMBER]@[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK227032cab. Max-Forwards: 70. Content-Length: 341. To: sip:[TO_NUMBER]@[EMAIL PROTECTED]. From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. ... # U 2008/12/19 12:24:24.374544 YYY.YYY.YYY.YYY:5060 -> XXX.XXX.XXX.XXX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK227032cab;rport=5060. To: sip:[TO_NUMBER]@[EMAIL PROTECTED];tag=b802e73ab92fec121a4f880263d541a3.7575. From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. ... # U 2008/12/19 12:24:24.407646 XXX.XXX.XXX.XXX:5060 -> YYY.YYY.YYY.YYY:5060 ACK sip:[TO_NUMBER]@[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK227032cab. Max-Forwards: 70. Content-Length: 0. To: sip:[TO_NUMBER]@[EMAIL PROTECTED];tag=b802e73ab92fec121a4f880263d541a3.7575. Route: <sip:[EMAIL PROTECTED]:5060;lr>. .... # U 2008/12/19 12:24:35.667448 XXX.XXX.XXX.XXX:5060 -> YYY.YYY.YYY.YYY:5060 INVITE sip:[TO_NUMBER]@[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK3362d177f. Max-Forwards: 70. Content-Length: 341. To: sip:[TO_NUMBER]@[EMAIL PROTECTED]. From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. Route: <sip:[EMAIL PROTECTED]:5060;lr>. ... U 2008/12/19 12:24:35.692576 YYY.YYY.YYY.YYY:5060 -> ZZZ.ZZZ.ZZZ.ZZZ:5060 INVITE sip:[TO_NUMBER]@ZZZ.ZZZ.ZZZ.ZZZ:5060;transport=udp SIP/2.0. Record-Route: <sip:[TO_NUMBER]@[EMAIL PROTECTED]:5060;ftag=72a0c76ae3d477c;lr=on>. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK32dd.6cc062c5.0. Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bK3362d177f. Max-Forwards: 69. Content-Length: 341. To: sip:[TO_NUMBER]@[EMAIL PROTECTED]. From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. ... U 2008/12/19 12:24:35.731432 ZZZ.ZZZ.ZZZ.ZZZ:5060 -> YYY.YYY.YYY.YYY:5060 SIP/2.0 100 Trying. Call-ID: 1c00f084ccd0232ad5ab638dc6a7ccfd@[EMAIL PROTECTED]. CSeq: 1965370819 INVITE. From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. .... U 2008/12/19 12:24:35.849714 ZZZ.ZZZ.ZZZ.ZZZ:5060 -> YYY.YYY.YYY.YYY:5060 SIP/2.0 183 Session Progress. Call-ID: 1c00f084ccd0232ad5ab638dc6a7ccfd@[EMAIL PROTECTED]. CSeq: 1965370819 INVITE. From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. To: sip:[TO_NUMBER]@[EMAIL PROTECTED];tag=86f125f633683d0. ... # U 2008/12/19 12:24:35.849961 YYY.YYY.YYY.YYY:5060 -> XXX.XXX.XXX.XXX:5060 SIP/2.0 183 Session Progress. Call-ID: 1c00f084ccd0232ad5ab638dc6a7ccfd@[EMAIL PROTECTED]. CSeq: 1965370819 INVITE. From: sip:[FROM_NUMBER]@[EMAIL PROTECTED];tag=72a0c76ae3d477c. To: sip:[TO_NUMBER]@[EMAIL PROTECTED];tag=86f125f633683d0. Can anybody help me to trobleshoot the issue? Thanks in Advance!! -- Krunal patel

2 1

[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 19 Dec '08

19 Dec '08

I will be out of the office starting 12/18/2008 and will not return until 12/29/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

1 0

Re: [Kamailio-Users] Kamailio in Red Hat 4 with mysql and carrierroute
by Juan Asencio 19 Dec '08

19 Dec '08

> On Monday 08 December 2008, Juan Asencio wrote: >> Thank you for your fast response. So after I compile them, the files >> carrierroute.so should appear on the directory of modules? >> >> Because I thought did it right, but I can't see the carrierroute.so >> anywhere. So I'm not sure if I did it right :( > > Hi Juan, > > you need to edit the exclude_modules parameter in the Makefile, and remove > the > db_mysql and carrierroute module. After a new make; make install this > modules > should also be available. > > Cheers, > > Henning > Hi Henning, now the modules are available on the kamailio.cfg file, but when I run the command "kamctl start" I got the following error: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed But, if comment on the .cfg file the carrierroute.so line, Kamailio is able to start. Do you have any suggestions about what I'm doing wrong? Cheers, Juan.-

2 2

[Serusers] database query
by Netadmin 19 Dec '08

19 Dec '08

Hello, Is there a function i can use to check if a user exist in subscriber table? Something similar to lookup? thanks

2 1

[Serusers] sip register problem SER behind NAT
by WANG Zhenzhong 19 Dec '08

19 Dec '08

I set the SER behind a NAT. The SER’s private IP is 192.168.9.88 The NAT’s public IP is 122.194.106.139 I use the eyebeam to register to SER from internet via the NAT address. But when the SER receive the register message, it find the IP(122.194.106.139) in “To” & “from” line is different from the local IP(192.168.9.88), So the SER forward the register message and the register is failed. How solve this problem. Thanks. 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 ============================================

2 4

[Serusers] the language of ser.cfg
by WANG Zhenzhong 19 Dec '08

19 Dec '08

I want to change the ser.cfg. Such as I want to define a new variable, and assign a string, etc But I don’t know which language the ser.cfg use. Is there any document to introduce the expression and language of ser.cfg? THANKS 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 ============================================

1 0

[Kamailio-Users] Is there a way to detect use_media_proxy called or not?
by Krunal Patel 18 Dec '08

18 Dec '08

Hi, Does anybody know how to detect whether use_media_proxy is called? What I want to do is: Do end_media_session for bye or cancel if & only if use_media_proxy has been called during invite or reply route. Is it possible? If it is then please let me know how to implement it? Thanks in advance -- Krunal Patel

4 4

[Serusers] SER with FreeRADIUS
by Robert Borz 18 Dec '08

18 Dec '08

Hi guys, we're using ser 0.9.7 on Debian/Lenny together with FreeRADIUS. Authentication against the RADIUS already works basically for our sip clients. Now we want to make the contact information persisting restarts of SER. Is there a way to store the contact information for the registered sip clients in the radius database or must an additional database, e.g. mysql or postgres, used for that? Regards, Robert ____________________________________________________________________ Psssst! Schon vom neuen WEB.DE MultiMessenger gehört? Der kann`s mit allen: http://www.produkte.web.de/messenger/?did=3123

1 0

Re: [Kamailio-Users] openser and asterisk auth problem.
by BERGANZ François 18 Dec '08

18 Dec '08

I want to authenticate in openser some users which are in asterisk conf (to do load balancing with another asterisk). If I delete the user from the asterisk config and insecure=invite for the openser, asterisk accept the call, but, I want that asterisk see that it is that user which is doing the call. Cordialement, BERGANZ François http://www.acropolistelecom.net  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Andrew V.Statsenko [mailto:alter@siptalk.ru] Envoyé : jeudi 18 décembre 2008 11:00 À : BERGANZ François Objet : ***SPAM*** Re: [Kamailio-Users] openser and asterisk auth problem. В Чтв, 18/12/2008 в 10:31 +0100, BERGANZ François пишет: > -----I have that error: > <--- SIP read from UDP://192.168.1.156:5060 ---> > INVITE sip:33662971130@192.168.1.156 SIP/2.0 > Record-Route: <sip:192.168.1.156;lr=on;ftag=0c46027a> > Via: SIP/2.0/UDP 192.168.1.156;branch=z9hG4bK20d2.de427962.0 > Via: SIP/2.0/UDP 192.168.1.60:43490;branch=z9hG4bK-d8754z-c15d2350d0742a11-1---d8754z-;rport=43490 > Max-Forwards: 70 > Contact: <sip:6833211245@192.168.1.60:43490> > To: "33662971130"<sip:33662971130@192.168.1.156> > From: "6833211245"<sip:6833211245@192.168.1.156>;tag=0c46027a > Call-ID: YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc. > CSeq: 2 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > Content-Type: application/sdp > User-Agent: X-Lite release 1100l stamp 47546 > Content-Length: 231 [..] > <-------------> > --- (14 headers 10 lines) --- > Sending to 192.168.1.156 : 5060 (no NAT) > Using INVITE request as basis request - YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc. > Found user '6833211245' for '6833211245' This is the key trouble in your configuration: Asterisk trying to authenticate this call as call from _user_ (INVITE->401->INVITE+auth), but not as _peer_. Will you describe us where you want to authenticate you users : - at the OpenSER frontend (BTW, I don't see auth headers in first INVITE) - or in the Asterisk backend ? Which service logic you are building ? -- WBR, Alter

3 4

Re: [Kamailio-Users] serMyAdmin database connection
by Flavio Goncalves 18 Dec '08

18 Dec '08

Message: 1 Date: Wed, 17 Dec 2008 16:29:56 +0200 From: TCB <tawandac(a)gmail.com> Subject: [Kamailio-Users] serMyAdmin error To: users <users(a)lists.kamailio.net> Message-ID: <f9db9e000812170629v34e6e1edr2a472912427c78eb(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" I'm trying to install serMyAdmin using a remote mysql DB. I keep getting the following error. Caused by: org.springframework.jdbc.CannotGetJdbcConnectionException: Could not get JDBC Connection; nested exception is com.mysql.jdbc.exceptions.jdbc4.CommunicationsException: Communications link failure I changed context.xml to point to the remote DB. Where else should i change DB the credential -- TC Please try to connect to your database using mysql command line, to check if your permissions are ok. I will verify the JDBC connection with a remote database. Regards, Flavio E. Goncalves

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