- sr-users - kamailio.org

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

19 Feb '09

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

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

19 Feb '09

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

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

19 Feb '09

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

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

19 Feb '09

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

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

19 Feb '09

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

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

19 Feb '09

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

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

19 Feb '09

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

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

19 Feb '09

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

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

18 Feb '09

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

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

18 Feb '09

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

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

17 Feb '09

Hello, The 1.0.1 release of SEMS, the SIP Express Media Server [0], is now available for download [1]. The 1.0.1 release contains all fixes applied to the 1.0 branch, and should be a drop-in replacement for anyone still using 1.0.0. Update is recommended to everyone who is still using 1.0.0-rcX or any later version from the 1.0 branch. Debian packages for etch/64 are also available [2]. Best Stefan Sayer [0] http://iptel.org/sems/ [1] http://ftp.iptel.org/pub/sems/sems-1.0.1.tar.gz [2] http://ftp.iptel.org/pub/sems/1.0/1.0.1/packages/debian Apologies if you receive multiple copies of this message. -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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

17 Feb '09

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

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

17 Feb '09

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

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

17 Feb '09

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

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

17 Feb '09

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

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

16 Feb '09

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

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

16 Feb '09

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

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

16 Feb '09

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

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

16 Feb '09

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

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

16 Feb '09

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

1 0

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

14 Feb '09

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

3 3

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

14 Feb '09

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

4 7

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

13 Feb '09

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

3 6

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

13 Feb '09

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

4 12

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

12 Feb '09

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

3 12

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

12 Feb '09

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

3 3

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

12 Feb '09

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

3 2

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

12 Feb '09

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

2 2

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

12 Feb '09

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

2 1

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

11 Feb '09

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

2 1

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

11 Feb '09

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

3 2

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

11 Feb '09

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

1 0

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

11 Feb '09

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

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

10 Feb '09

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

1 0

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

10 Feb '09

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

1 0

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

10 Feb '09

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

2 2

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

10 Feb '09

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

2 1

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

09 Feb '09

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

1 0

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

06 Feb '09

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

3 6

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

06 Feb '09

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

2 1

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

06 Feb '09

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

2 1

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

06 Feb '09

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

3 2

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

05 Feb '09

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

3 12

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

05 Feb '09

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

2 7

[Serusers] preload_udomain error
by caio 05 Feb '09

05 Feb '09

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

1 1

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

05 Feb '09

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

2 1

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

04 Feb '09

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

2 4

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

04 Feb '09

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

4 4

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

04 Feb '09

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

1 0

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

04 Feb '09

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

1 0

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

04 Feb '09

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

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

03 Feb '09

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

2 4

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

03 Feb '09

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

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

03 Feb '09

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

2 1

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

02 Feb '09

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

1 0

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

02 Feb '09

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

2 3

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

02 Feb '09

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

2 2

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

02 Feb '09

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

1 1

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

31 Jan '09

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

3 4

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

30 Jan '09

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

1 1

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

30 Jan '09

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

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

29 Jan '09

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

1 0

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

29 Jan '09

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

1 0

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

29 Jan '09

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

1 0

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

29 Jan '09

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

3 4

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

28 Jan '09

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

2 1

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

28 Jan '09

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

2 2

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

28 Jan '09

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

3 2

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

27 Jan '09

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

4 3

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

27 Jan '09

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

2 3

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

27 Jan '09

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

2 1

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

26 Jan '09

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

2 1

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

26 Jan '09

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

2 3

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

26 Jan '09

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

2 1

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

25 Jan '09

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

2 2

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

24 Jan '09

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

4 6

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

23 Jan '09

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

3 8

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

23 Jan '09

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

2 9

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

23 Jan '09

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

2 7

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

22 Jan '09

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

2 1

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

22 Jan '09

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

3 6

[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

3 3

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

22 Jan '09

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

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

22 Jan '09

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

3 2

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

21 Jan '09

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

1 0

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

21 Jan '09

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

1 0

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

21 Jan '09

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

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

21 Jan '09

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

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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

2 2

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

20 Jan '09

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

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

20 Jan '09

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

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

20 Jan '09

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

2 1

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

19 Jan '09

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

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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…

2 1

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

19 Jan '09

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

2 1

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

19 Jan '09

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

1 0

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

19 Jan '09

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

2 1

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

17 Jan '09

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

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

16 Jan '09

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

2 3

[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

5 17

[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

3 4

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

16 Jan '09

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

2 3

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

16 Jan '09

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

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

16 Jan '09

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

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

15 Jan '09

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

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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. ============================================================================================================================

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

12 Jan '09

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

1 0

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

12 Jan '09

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

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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 >> >> >

2 5

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

18 Dec '08

Hello, I have : ASTERISK-----OPENSER-----SOFTPHONE My softphone auth to openser (with username/password of asterisk sync with a database) Openser forward the INVITE to asterisk and asterisk return Unautorize! I tried to have the same realm to try de have the same auth for asterisk and openser But without success. In fact, I need to auth to openser(with asterisk config user in the database) and forward to asterisk the INVITE. Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] Use Carrierroute to rewrite from_uri based on prefix table
by rod 18 Dec '08

18 Dec '08

Hi all, I've been able to rewrite R_URI based on prefix matching in the carrierroute module, using: cr_route("default", "0", "$rU", "$rU", "call_id") I used this to normalize the ruri to E164: ex: in france, the country code is +33, if someone dials "01 10 10 10 10", I strip the first 0 and prefix with 33 so that I get 331 10 10 10 10 (space are there for convenience :p) What I'd like to do is to use the carrierroute module to do the same but for from uri: ex: I receive a call on my phone with a caller using e164 (from_uri: 331 20 20 20 20) and I'd like to present this number as 01 20 20 20 20 to the callee. I added a prefix 331 in the carrierroute tables. I tried this: - cr_route("default", "0", "$fU", "$fU", "call_id") - cr_route("default", "0", "$fU", "$fU", "from_uri") - cr_route("default", "0", "$fU", "$rU", "call_id") - cr_route("default", "0", "$fU", "$rU", "from_uri") but I've been unsuccessful with this. Is there a way to achieve from_uri translation using the carrierroute module. Regards, rod

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[Kamailio-Users] nat problems
by Ricky Gutierrez 17 Dec '08

17 Dec '08

Hi list, I want to comment some problems of nat that I have, I have two connected remote UAC in a same location UAC 202 REMOTE A UAC 123 REMOTE A if the UAC 123 calls to UAC 202 there is not bidirectional audio UAC 204 REMOTE B UAC 122 REMOTE C if the UAC 122 calls to UAC 202 if there is bidirectional audio some help like I can solve this problem?, the remote clients in site A they are behind a nat symetrics I am using rtpproxy version 1.1 and openser 1.3.2 log sip U +2.975758 190.212.49.86:58944 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152:5060;branch=z9hG4bK74f3.13c6ec23.0 Via: SIP/2.0/UDP 192.168.10.37:5060;rport=63451;received=190.212.49.86;branch=z9hG4bK33571797d7d31cb0 Record-Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:202@192.168.10.40:5064;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=202 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 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000475 190.184.22.152:5060 -> 190.212.49.86:63451 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.37:5060;rport=63451;received=190.212.49.86;branch=z9hG4bK33571797d7d31cb0 Record-Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:202@190.212.49.86:58944;transport=udp;nat=yes> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 234 P-hint: onreply_route|force_rtp_proxy P-hint: Onreply-route - fixcontact v=0 o=202 8000 8000 IN IP4 192.168.10.40 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35036 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=nortpproxy:yes # U +0.241543 190.212.49.86:63451 -> 190.184.22.152:5060 ACK sip:202@190.212.49.86:58944;transport=udp;nat=yes SIP/2.0 Via: SIP/2.0/UDP 192.168.10.37:5060;branch=z9hG4bKa26522558c582a68 Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Contact: <sip:123@192.168.10.37:5060;transport=udp> Supported: path Proxy-Authorization: Digest username="123", realm="190.184.22.152", algorithm=MD5, uri="sip:202@190.184.22.152", nonce="49485f8958ba6dc4e39ee5893d15ea3e60ae19ad", response="afb5db07f3a90d4cfc6ea483fd120404" Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 ACK User-Agent: Grandstream BT200 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.000334 190.184.22.152:5060 -> 190.212.49.86:58944 ACK sip:202@190.212.49.86:58944;transport=udp SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK74f3.13c6ec23.2 Via: SIP/2.0/UDP 192.168.10.37:5060;rport=63451;received=190.212.49.86;branch=z9hG4bKa26522558c582a68 From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Contact: <sip:123@190.212.49.86:63451;transport=udp;nat=yes> Supported: path Proxy-Authorization: Digest username="123", realm="190.184.22.152", algorithm=MD5, uri="sip:202@190.184.22.152", nonce="49485f8958ba6dc4e39ee5893d15ea3e60ae19ad", response="afb5db07f3a90d4cfc6ea483fd120404" Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 ACK User-Agent: Grandstream BT200 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.383479 190.184.22.152:5060 -> 190.212.49.86:58944 OPTIONS sip:190.212.49.86:58944 SIP/2.0 Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=d7259e51 To: sip:190.212.49.86:58944 Call-ID: 1f9f65c1-fcccdef2-674a(a)190.184.22.152 CSeq: 1 OPTIONS Content-Length: 0 # U +0.173847 190.212.49.86:58944 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=d7259e51 To: sip:190.212.49.86:58944;tag=15c74eb7e1a2ae92 Call-ID: 1f9f65c1-fcccdef2-674a(a)190.184.22.152 CSeq: 1 OPTIONS User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:202@192.168.10.40:5064;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 # U +0.826048 190.184.22.152:5060 -> 190.212.35.160:50273 OPTIONS sip:190.212.35.160:50273 SIP/2.0 Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=e7259e51 To: sip:190.212.35.160:50273 Call-ID: 1f9f65c1-0dccdef2-774a(a)190.184.22.152 CSeq: 1 OPTIONS Content-Length: 0 # U +0.192726 190.212.35.160:50273 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=e7259e51 To: sip:190.212.35.160:50273;tag=c2c330e1c7bae7ba Call-ID: 1f9f65c1-0dccdef2-774a(a)190.184.22.152 CSeq: 1 OPTIONS User-Agent: Grandstream GXP2000 1.1.6.16 Contact: <sip:203@192.168.0.16:5060;transport=udp;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 # U +1.807039 192.168.11.1:5060 -> 192.168.11.30:5060 OPTIONS sip:192.168.11.30:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.1:5060;branch=0 From: sip:pinger@190.184.22.152;tag=f7259e51 To: sip:192.168.11.30:5060 Call-ID: 1f9f65c1-1dccdef2-974a(a)192.168.11.1 CSeq: 1 OPTIONS Content-Length: 0 regards rickygm

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[Kamailio-Users] simple call between 2phones
by BERGANZ François 17 Dec '08

17 Dec '08

Hello, I know kamailio, but, I dont know how to do : Call between 2phone which are registered ! Know you how to do ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] serMyAdmin error
by TCB 17 Dec '08

17 Dec '08

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 -- TC

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[Kamailio-Users] failed to request mysql lock ()
by Brian Del Shasta 16 Dec '08

16 Dec '08

Is this an issue? Dec 16 14:47:45 fndipt03 cdrtool[10738]: Database error: failed to request mysql lock () Dec 16 14:47:45 fndipt03 cdrtool[10738]: Unlock ser_radius:radacct Dec 16 14:47:45 fndipt03 cdrtool[10738]: Error in unLockNormalization() The only problem I have is I am unable to get the pricing to work, but I think that may be unrelated... Thanks

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[Kamailio-Users] Kamailio v1.4.3 Released
by Daniel-Constantin Mierla 16 Dec '08

16 Dec '08

Hello, a new release in 1.4 series is out. Kamailio 1.4.3 is based on the latest version of branch 1.4, including many fixes in code and documentation, therefore those running 1.4.0, 1.4.1 or 1.4.2 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.4.3/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.4.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4 kamailio Sources for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/branches/1.4/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.4.x/ What is new in 1.4.x release series is summarized in the announcement of v1.4.0: http://www.kamailio.org/mos/view/KAMAILIO-v1.4.0-Released Note: Kamailio is the new name of OpenSER project. First version under Kamailio name was 1.4.0. Older versions will continue to use OpenSER name. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] memory leak debugging
by joy yue 16 Dec '08

16 Dec '08

Hi, I was trying to debugg if I hit a memory leak issue. I delete "-DF_MALLOC" line and insert a "-DDBG_QM_MALLOC" line of DEFS string in Makefile.def. I was able to compile openser with new Makefile.def. But when running openser, I get error, fm_free() is not defined, in many modules. And there is a flag mismatch between openser core and modules compilation. Core compilation has DDBG_QM_MALLOC flag, but module compilation still uses F_MALLOC instead of DDBG_QM_MALLOC. Is there anything else I need to change in Makefile.defs? Thanks, -Joy

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Re: [Kamailio-Users] Kamailio in Red Hat 4 with mysql and carrierroute
by Henning Westerholt 16 Dec '08

16 Dec '08

On Tuesday 16 December 2008, Juan Asencio wrote: > [..] > I need to test the carrierroute module in Kamailio and I been reading the > documentation about it. > > http://www.kamailio.org/docs/modules/devel/carrierroute.html#id2506610 > > But this something entirely new for me. So I have a couple of questions > that I hope you can answer or guide me to where I could found an answer. > > I need to test it with two Asterisk gateways. So I guess I need to edit my > kamailio.cfg file as the example on "Example 1.19. Configuration example - > module configuration" My question is: On prefix 49 and prefix NULL, are > these the two routes available? If the prefix is 49, so the call go > through proxy1.localdomain or proxy2.localdomain and if the prefix is NULL > it would go through register1.localdomain or register2.localdomain? > > It is proxy and register the gateways? Hi Juan, please write this questions to the user list. The config in question is probably: domain proxy { prefix 49 { ... } } domain register { prefix NULL { ... } } This defines two "domains" in the default carrier tree. The 'proxy' domain will be used if you specify 'proxy' in the cr_route calls, the same applies to the 'register' case. So if you want to distribute traffic between two asterisk, just use one domain, and a empty (NULL) prefix. See the following example config: domain foobar { prefix 49 { max_targets = 2 target box1.localdomain { prob = 0.500000 hash_index = 1 status = 1 comment = "test target 1" } target box2.localdomain { prob = 0.500000 hash_index = 2 status = 1 comment = "test target 2" } } prefix NULL { max_targets = 1 target box3.localdomain { prob = 1.000000 hash_index = 1 status = 1 comment = "test target 3" } } } This way all traffic with the prefix '49' will be equally distributed between box1 and box2, and all other prefixes will be routed to box3. Hope this is now more clear. > Could you suggest me what to read in order to get a better understanding > of this module? You probably need to experiment a bit. If you increase the log level to INFO, then the module will report what it does, increase to DEBUG to get even more informations, e.g. about domain searching. Take a look to the mailing list archives, if you get stuck just ask at the user list. Cheers, Henning

1 0

[Kamailio-Users] k/o xlog and pv modules for srouter
by Daniel-Constantin Mierla 16 Dec '08

16 Dec '08

Hello, I have uploaded xlog and (stripped) pv Kamailio/Openser modules to sip-router.org site at: http://sip-router.org/pub/tmp/miconda/ If you are willing to test them, then just get the branch "master" from git and merge with "daniel/pv". Details about GIT repository: http://sip-router.org/2008/11/18/git-repository-online/ The pv module does not include yet the pseudo-variables and transformations dependent of K/O extensions (e.g., branch, script flags, ...). Probably one important to decide upon is the AVP (to start a new discussion about it separately). The ones missing now will be gradually activated as dependent code is sorted out. Testing and feedback is welcome. Andrei is doing heavy work to improve the config script interpreter for string and arithmetic operations, you can track the commits at: http://lists.sip-router.org/pipermail/sr-dev/ Once work is finished and the developers' branches are merged, we can mark an important milestone - K/O pseudo-variables working in sip-router config file, opening full functionality for large set of K/O modules and clean compatibility of config files for many typical deployments. SIP-Router is able to load now both types of modules: ser and k/o. DB APIs from both projects were tested as library, from K/O point of view mi and statistics API are the extensions yet to be made libraries for sip-router. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] CDRTool --> CDRs= signalling information ok but no Rating informations
by BERGANZ François 16 Dec '08

16 Dec '08

Hello, I could install CDRTool. I have in CDRs signallings informations but dont have Rating information. Have you an idea? Id Start time Sip Proxy SIP caller In SIP destination Out Dur Price KBIn KBOut Status Codec <http://192.168.1.150/CDRTool/callsearch.phtml> 1N 2008-12-16 11:23:24 2000(a)192.168.1.155 33170725015@192.168.1.15100:19 Ok (200) Signalling information <http://192.168.1.150/CDRTool/callsearch.phtml?cdr_source=ser_radius&cdr_tab le=radacct&order_by=RadAcctId&order_type=DESC&begin_datetime=1229420460&end_ datetime=1229468100&maxrowsperpage=15&action=search&call_id=YjUwMzg1OGVlYzU4 M2QwNzNkZjQ4ZjNmYzUyZTczNGM.> Click here to show only this call id Call id: YjUwMzg1OGVlYzU4M2QwNzNkZjQ4ZjNmYzUyZTczNGM. <javascript:void(null);> Click here to see the SIP trace for this call From/to tags: 636e0825/as5d7d6d2a Start time: 2008-12-16 11:23:24 Europe/Amsterdam Stop time: 2008-12-16 11:23:43 Method: Invite from 192.168.1.60:39366 From: 2000(a)192.168.1.155 Domain: 192.168.1.155 To (dialed URI): 33170725015(a)192.168.1.155 Canonical URI: 33170725015(a)192.168.1.151 Next hop URI: 33170725015(a)192.168.1.151 Billing Party: 2000(a)192.168.1.155 Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] documenting pseudo-variables and transformations
by Daniel-Constantin Mierla 16 Dec '08

16 Dec '08

Hello, today the PV&T are documented in several places: - core PV&T were on dokuwiki page - PV exported by modules in the README As of now, all PV&T from core were moved in pv module, but they are sill documented on dokuwiki page. However, we should get to a consistent way. My suggestion is to document PV&T in docuviki, mentioning which module to load to become available. We keep a single page a reference, being easier to find all the PV&T, rather than going over all modules' README. I think user experience is more straightforward. The module readme should maintain a small section, listing the names of PV&T plus a link to dokuwiki page. Any other opinions? I am just about to document the new $T_rpl(pv) and $T_req(pv) pseudo-variables from TM, so I am waiting to do it in the proper place: http://lists.kamailio.org/pipermail/devel/2008-December/017108.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Simple Proxying?
by Jeffrey Ollie 16 Dec '08

16 Dec '08

Hello, I'm a newbie with OpenSER/Kamailio so please bear with me. I'm trying to use OpenSER 1.3.4 and rtpproxy to connect two IP PBXs. The two PBXs (and the phones they serve) cannot talk directly to each other, they need to go through my OpenSER system. One system is a Cisco CallManager cluster (v5.1.3) which I administer and the other is a Mitel system run by another organization. I have a CentOS box running OpenSER and rtpproxy with two interfaces one on each network. Routing on the CentOS box appears correct as I can ping everything I expect to. I've gotten OpenSER configured so that SIP messages are sent between the PBXs, but the SDPs aren't getting rewritten properly and the RTP isn't getting passed through the rtpproxy. I've attached my config so far, can anyone point out where I've gone wrong or point me to some example configs that might help me out? All of the example configs that I've found so far seem to be oriented towards SIP endpoints. -- Jeff Ollie "You know, I used to think it was awful that life was so unfair. Then I thought, wouldn't it be much worse if life were fair, and all the terrible things that happen to us come because we actually deserve them? So, now I take great comfort in the general hostility and unfairness of the universe." -- Marcus to Franklin in Babylon 5: "A Late Delivery from Avalon"

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[Kamailio-Users] avp script flags
by Daniel-Constantin Mierla 16 Dec '08

16 Dec '08

Hello, while adapting the PV API, I got into the AVP script flags? Is anyone still using them? If yes, for what purpose, maybe we can figure out a nicer way and drop these flags. We have to find the solution to make AVPs from kamailio/openser and ser compatible in the new sip-router.org project. Just for those not being sure what I am talking about, avp names can have flags after the name type specifier, like: $avp(i3:23) - meaning that flags 1 and 2 (bitmask ...0011) are set for that avp. As I haven't seen any related discussions for quite some time, the functionality might not be updated all over the code. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] tracking modules
by Daniel-Constantin Mierla 16 Dec '08

16 Dec '08

Hello, as we have a big number of modules and we did updates like renaming, merging, ..., I have created a dokuwiki page to track a module. Now it is a table that should show for each module the date (release) when it was introduced, current status and description about that. Feel free to enhance the format, the page is located at: http://www.kamailio.org/dokuwiki/doku.php/modules:status Now it is just a skeleton. I hope all of you will contribute there. You can add to description anything you believe is relevant for that module. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Configuring Kamailio with Asterisk gateways
by Juan Asencio 15 Dec '08

15 Dec '08

Hi all, I'm trying to configure my kamailio sip server with two asterisk gateways. I have look for some tutorials, but the one in Kamailio website have me a little confuse, because the files content is different. I guess it must be the .cfg file from Openser the one that is describe on the website. http://kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration Could any one suggest me some tutorials, to see how the kamailio.cfg file should be edited? Best regards, Juan.-

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[Kamailio-Users] Contact of reply for SUBSCRIBE
by Andreas Granig 15 Dec '08

15 Dec '08

Hi all, Kamailio sends a Contact set to modparam "server_address" in the "presence" module when replying to SUBSCRIBE. Now I've seen in some clients (e.g. eyebeam) and libs (e.g. eXosip) that they send subsequent SUBSCRIBEs to this contact. In case of eXosip, it's for automatically refreshing the subscription. Problem is, that I can't tell what to put into "server_address" to handle this correctly, since just putting in the domain of my server (like "sip:my.domain") doesn't work, and putting in some generic URI like "sip:server@my.domain") doesn't make sense also, since the user doesn't exist. So, how are you supposed to handle this? Any thoughts? Andreas

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[Kamailio-Users] serial forking : is hash table autoexpire working ?
by Aurelien Grimaud 15 Dec '08

15 Dec '08

Hi, I am using openser to route requests towards differents Interactive Voice Response (IVR). For this, and to manage failures, maintenance a flexible way, I use serial forking. For each requests (after loose routing process), all possible destinations are retrieved from 'location' with a modified lookup. The first one is pushed to DURI, the other ones are stored in AVP. An on_failure_route is set to get next destination if any. fr_timer is 5 secs fr_inv_timer is 120 secs. Therefore, if an IVR does not response at all within 5 secs, the serial forking mechanism is triggered. Any subsequent message *MUST* go to same IVR, this is the record_route/loose_route processing, and everything is ok for ACK or BYE. My problem is for requests that does not match loose_routing. - What about the initial INVITE being retransmitted ? - What about a CANCEL ? In those cases, the request goes to another IVR which handles it genuinely. For INVITE retransmission, the calling uac will get two different answers. If lucky, one will be canceled. For CANCEL, the message will be rejected by all destinations but the one which expects it (if lucky). To avoid this, I used dynamic avp to set nexthop for a specific callid, as soon as nexthop was chosen. These avp were stored in database and reloaded from ... This is really slowing down the system for a little annoyance ... With kamailio 1.5.0 I use new module hash table to store next hop. nexthop is cleaned on BYE, CANCEL and failures. To handle runaway calls, the autoexpire parameter of hash table is a great thing. Unfortunately It does not seem to do the job. All entries stays indefinitely in table (as shown in sht_print) Do you think these serial forking issues (re-INVITE, CANCEL) and all the work around (nexthop hash table) it need is useful ? Anyone want to share its serial forking management ? What about the autoexpire parameter for hash table ? Is it ok or did I miss something ? loadmodule "htable.so" modparam("htable", "htable", "nexthop=>size=4;autoexpire=120;") Aurelien

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[Serusers] Redirect server without "305 Use Proxy"?
by Rick van Rein 14 Dec '08

14 Dec '08

Hello, I am trying to setup a redirect proxy that pre-parses SIP before it gets to the telco. It can do things like lookup speeddial-codes and map them to either a phone number or a SIP-address. My first thought was to use "305 Use Proxy" to tell phones to move on to the proxy of the telco. But 305 is discouraged on iptel.org as it is not properly standardised. 1. Am I correct in assuming that it is best to use "302 Moved Temporarily"? Using this, I would need to set the Contact: field with address(es) of new servers to appraoch. 2. What is the designated method to set Contact: in preparation of "302"? Since there may be more than one SIP-server at the provider, it is good to be able to create multiple Contact: headers. Behind the @ could be server names and ports (from SRV lookup) or IP numbers (from A lookup). 3. How to make DNS lookups for proxy servers with Ser? 4. How to fold multiple DNS lookup results onto multiple Contact: headers? I wonder if this is an existing facility in Ser, or if I should extend Ser for doing this? I hope someone on this list can help me move forward. Thanks! -Rick

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