- sr-users - kamailio.org

AUTOREPLY Re: Problemdebian squeze uac module load failed
by strobel＠hfo-telecom.de 21 Dec '12

21 Dec '12

******************* AUTOREPLY VON miconda(a)gmail.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 27.12.2012 bis einschließlich 01.01.2013 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Brunner oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hello, do you get another error message before? If not, set debug=3 in kamailio.cfg and run again. Then look at the logs to see if there is another message giving hints about what is wrong. Probably some mis-configured parameter. Cheers, Daniel On 12/21/12 4:38 PM, Thomas Hoellriegel wrote: > Hi all, i using debian squeeze. > I installled kamailio from the package system. > I set in the kamailio.cfg file: > loadmodule "uac.so" > I become the error: > Dec 21 16:37:35 1a-7567 > /usr/sbin/kamailio[3015]: ERROR: <core> [sr_module.c:939]: init_mod(): > Error while initializing module uac > (/usr/lib64/kamailio/modules_k/uac.so) > > Can you help plese? > thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste für blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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RPMs for v3.3.3
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, for the records, the RPMs for v3.3.3 are now available (32b and 64b: CentOS 5 and 6, RedHat 5.0 and 6.0, OpenSUSE 11.4 and 12.1, Fedora 15 and 16). They are hosted by opensuse build service, links specific per distribution are listed at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] [sr-dev] kamailio-ser integration - db modules
by Ovidiu Sas 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:20 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> What's the difference between the db1 and db2 interface? > > It is not the case here, modules_k/ldap does not have any relation to > database interface. Any database module db_[DatabaseType] implements an API. SER was using one API and opensips/kamailio a different API (and therefore today we have version 1 and 2). All db modules in the stable release are supporting both APIs. The new db ldap module has support only for the SER API version. The API is documented in each lib/srdb[1|]/*.h header files. The ldap module from ser is completely different than ldap module from kamailio (despite the fact that both are using the same name). It's the same with the dialog module. Check the README file for each. Hope this brings a little bit of light :) -ovidiu

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Re: [SR-Users] [sr-dev] kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

On 12/20/12 4:13 PM, Ovidiu Sas wrote: > On Thu, Dec 20, 2012 at 9:39 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >>>> - nathelper - some extra functionality, not sure if can be kept >>>> completely >>> Maybe Andreas can look into this, as there is a lot of work going on with >>> nathelper and the new rtpproxy anyways. >> I think Ovidiu Sas looked at it when he split the rtpproxy out in a >> dedicated module. > IIRC, ping_contact whas the extra functionality in nathelper: > http://sip-router.org/docbook/sip-router/branch/master/modules_s/nathelper/… > I don't know how widely used is this functionality. > > Maybe we should have a separate thread per module (in user mailing > list to gather more imput) and see if it's worth merging the code or > use only the k version. I re-cc-ed the thread to users in case someone has comments to it. > > Also, on a separate note, I saw the we have a few db2_[module]. > I think it would make sense for these modules to rename them into > [module]_db[1|2]. > For example: ldap - we should have both versions under modules: > - ldap_db1 > - ldap_db2 > Just a suggestion ... The type of the two ldap modules are different, modules_k/ldap is a connector to ldap server from configuration file, offering possibility to do ldap search queries from config. The former modules_s/ldap (now db2_ldap) is a DB API v2 implementation driver module, so it can be used as a replacement for db_mysql (for example) when using some modules (such as db2_ops). I prefixed with db2_ to indicate that is not implementing DB API v1. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, most of the duplicated or ser-specific modules were sorted out at this time, next is the situation with the remaining ones. A) modules that will be renamed using a prefix 'uid_', because they have a database schema using unique id instead of username and domain: - auth_db => uid_auth_db - avp_db => uid_avp_db - domain => uid_domain - gflags => uid_gflags (not documented, but this module loads so called ser global avps) - uri_db => uid_uri_db B) modules to become obsolete with some work to be done - cpl-c - seems that giving a parameter with what user to be used for loading the cpl scripts will make k and s versions compatible - maxfwd - different names for config functions - nathelper - some extra functionality, not sure if can be kept completely - textops - seems to have some extra cfg functions - pike - has a rpc command - registrar/usrloc - avps stored per contact C) modules to become obsolete, most probably without further work - permissions - different structures/database table internally, but all should be achieved using k version (data migration might be required) D) modules that needs further investigations - rr - there seems to be some feature related to avps and record-route parameters/cookies - this module could be just renamed at the end (like rrs) E) Sample modules, not for real use - they can be kept in this directory (modules_s[ample]) - print - print_lib If anyone has comments regarding these remarks, reply here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Bug in uac_replace_from if used more than once?
by Daniel Tryba 20 Dec '12

20 Dec '12

Apparantly it should be simple to mangle the From in request: http://kamailio.org/docs/modules/stable/modules_k/uac.html#id2525538 The task at hand is use the prefix 131 to suppress callerid for this call: if($rU=~"^131[0-9]+$") { strip(3); append_hf("Privacy: id\r\n"); uac_replace_from("anonymous","sip:anonymous@anonymous.invalid:$rp"); uac_replace_to("","sip:$rU@$rd"); } Just before this replace there is an other uac_replace_from to rewrite an alphanumeric username to the default callerid for that device. If there are 2 uac_replace_from in the path, things go wrong like below. A uac_restore_from() before the second uac_replace_from doesn't change this behavior. device calls 1310123456789: U 10.0.34.226:1300 -> 10.0.35.226:5060 INVITE sip:1310123456789@kamailio.local:5060 SIP/2.0. From: <sip:anonymous@10.0.3.44:5060>;tag=26e5ce5850. To: <sip:1310123456789@kamailio.local:5060>. Kamailio forwards request to dispatcher but fails to properly replace the from uri: U 10.0.35.226:5060 -> 10.0.32.36:5060 INVITE sip:+31123456789@kamailio.local:5060 SIP/2.0. From: "anonymous" <sip: +31987654321@10.0.3.44:5060sip:anonymous@anonymous.invalid:5060>;tag=26e5ce5850. To: <sip:0123456789@kamailio.local>. Even though the debug log suggests it did the proper replacement: uac [replace.c:265]: value to store is is '<sip:anonymous@10.0.3.44:5060>' and len is '30' uac [replace.c:268]: Storing in FROM-AVP (for use in reply): '<sip:anonymous@10.0.3.44:5060>' with len '30' uac [replace.c:324]: uri to replace [sip:anonymous@10.0.3.44:5060] uac [replace.c:325]: replacement uri is [sip:anonymous@anonymous.invalid:5060] uac [replace.c:383]: encode is=<AAAAAAAAAAAAAAAAAABQXkFeV15BQUcUXF5AUWxpZDo1MDYw> len=48 uac [replace.c:265]: value to store is is '<sip:1310619024127@kamailio.local:5060>' and len is '54' uac [replace.c:271]: Storing in TO-AVP (for use in reply): '<sip:1310123456789@kamailio.local:5060>' with len '54' uac [replace.c:324]: uri to replace [sip:1310123456789@kamailio.local:5060] uac [replace.c:325]: replacement uri is [sip:0123456789@kamailio.local] uac [replace.c:383]: encode is=<AAAAAAEFAAkGAw0BAANxQlYuERJDCRxeHAoXWgYRXRdbAxtNHxwNCgdACxgubmw6NTA2MA--> len=72 Kamailio version: kamailio 3.3.2 (x86_64/linux) (the latest stable debian/squeeze package). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

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Changing METHOD
by Olle E. Johansson 20 Dec '12

20 Dec '12

Hi! I want to change METHOD on a SIP message from UPDATE to INVITE or REGISTER to INFO. Seems like $rm is read-only. Any other known way to change a requests method? /O

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Send reply with Reason header
by Mino Haluz 19 Dec '12

19 Dec '12

Hi, I know how to use sl_send_reply, but I would like to add "Reason:" header to the reply generated. Is there any command or variable for that? Thank you, Mino

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Video conference with many users
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, I want to do web (HTML5 + WebRTC) Sip client which can do the video conference with multiple users. Current release of SIPML <http://www.sipml5.org/> does 1 to 1 call. I have no idea of conference with many users. Is it the client that we need to modify to accept call and join the conference ? Do I need to send INVITE with extra parameters ? Please advice me. Best Regards, Roy.

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Question registering sip providers in kamailio and forward all?
by Thomas Hoellriegel 19 Dec '12

19 Dec '12

Hi all, I.m. a newbi in Kamailio. My problem: I.m. a blind user and i programmed a voicechat in Freeswitch. 3 have 2 sip providers. vodafone.de sipgate.at and freenet.de. The problem: i must register these proviers in kamailio. And forward all incoming calls to Freeswitch, and outgoing calls from freeswitch to kamailio. Dtmf modes are: auto rfc2833 and inband. Then i call: dorf(a)dorf.blindi.net i like to forward these calls to my freeswitch box. the kamailio ip ist: 217.172.180.108 and the freeswitchbox: 217.172.170.120 Can your help me please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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NAT Traversal issue
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client->LAN client web page call 2. LAN webrtc client -> 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy.

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New developer: Richard Good
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Richard Good - he is among those developing the IMS extensions for Kamailio (http://code.google.com/p/smile-ims-extensions/), to be merged in the next days to the GIT master branch. His git commit id is: richard.good My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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modules_s/timer review
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I was comparing modules_s/timer with modules_k/rtimer to spot the differences. modules_s/timer is working only with core timers (fast and slow timers), while the modules_k/rtimer can start own timer processes, or register a task to main core timer. Other differences ... modules_s/timer: - run on milisecond basis - option to enable/disable timer modules_k/rtimer - run either on second or micro-second basis - option to execute the many route blocks on same timer So, apart of enable/disable timer option, rtimer can do more than what timer module offer (mili-seconds can be run as 1000x micro-second). Enabling/disabling a timer can be workarounded from script, using a shared variable tested at begininning of rtimer route, like: route[RTIMER] { if($shv(rtimer)==0) return; ... } Enabling will mean setting $shv(rtimer)=1, either from other parts of the config or via MI/RPC commands. Now, both modules can be kept, no naming conflict, it is just a matter of code maintenance if we should keep timer module. Enabling/disabling timers can be added to rtimer in the future, but it is unlikely to happen before next major release (unless someone else does it). Is anyone here using modules_s/timer module aware of other differences comparing with rtimer? Looking for comments to keep or obsolete modules_s/timer module ... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Kamailio v3.3.3 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Planning next minor release - v3.3.3
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, I am considering releasing next minor version from branch 3.3 at the beginning of next week, Monday or Tuesday (Dec 17 or 18). For outstanding issues, please use the tracker or send to sr-dev mailing list. Developers that pushed fixed in the master branch should backport them in 3.3 before the next week starts. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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dlg_manage() and add_contact_alias() problem
by Pavel Miskov 18 Dec '12

18 Dec '12

Hi all, I want to end dialog (Kamailio sends BYE to both parties) that is lasting longer then some defined time. everything is working fine when using old fix_nated_contact function, but when using new add_contact_alias function kamailio sends BYE to private addresses. 1. "kamctl fifo dlg_list" output when using fix_nated_contact(): caller_contact:: sip:bob@PUBLIC_IP:50784 callee_contact:: sip:alice@PUBLIC_IP:64472 2. "kamctl fifo dlg_list" output when using add_contact_alias(): caller_contact:: sip:bob@10.2.5.206:10000 callee_contact:: sip:alice@10.2.5.205:20000 dlg_manag() doesn't store contact alias for caller and callee so Kamailio is sending BYE to private addresses. Am I doing something wrong or I can't use possibility of ending dialog with contact aliases? Thank you Pavel

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Support
by Murali Krish 18 Dec '12

18 Dec '12

Hi this is Murali krishnan i am final student of master degree in Dalhousie University, I am Working on my project, i m need of a sip server which can support H.264 codec video calls...I created an account in iptel sip account when i used the account it doesn't support video calls. If i use SER will this support video calls? -- *Regards Murali Krish*

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From sip phone to provider trunk
by andre second 18 Dec '12

18 Dec '12

Hi,I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that?Thank you!

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p_usrloc registration update problem
by Hans Goossens 18 Dec '12

18 Dec '12

Hi, I'm testing the p_usrloc module (Kamailio version 3.3.2, installed via ubuntu package) and it looks like it's working but for some reason a re-registration is causing a new entry being added to the distributed location table instead of updating the existing one. The kamailio configuration looks like this: #!define DBURL "mysql://openser:openserrw@localhost/openser" loadmodule "p_usrloc.so" modparam("p_usrloc", "db_mode", 3) modparam("p_usrloc", "write_db_url", DBURL) modparam("p_usrloc", "read_db_url", DBURL) modparam("p_usrloc", "write_on_db", 1) modparam("p_usrloc", "alg_location", 1) modparam("p_usrloc", "reg_db_table", "locdb") mysql> select * from locdb; +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | id | no | url | status | errors | failover | spare | rg | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | 1 | 1 | mysql://root:secret@172.16.10.131/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | | 1 | 2 | mysql://root:secret@172.16.10.132/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ When using the default usrloc functionality everything is working fine. The only difference I noticed is that the default usrloc is using MYSQL UPDATE to update the location entry and that p_usrloc is using MYSQL REPLACE. Regards Hans

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variable in rewritehost ?`
by Ali Jawad 17 Dec '12

17 Dec '12

Hi I am using slqops to get some user related info from location table, I want to use that info in rewritehost, however when I do use a variable in rewritehost I get an error "Unable to resolve variable". Is it possible to use a variable in rewritehost ? If not what is the workaround ? Thanks

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Dialog module not decrementing profile size
by Grant Bagdasarian 17 Dec '12

17 Dec '12

Hello Community, I've recently started to use the Dialog module to manage the amount of incoming calls from our SIP provider allowed to pass through by using profiles_with_values and setting the size of these profiles to a certain amount. Most of the time this works as it should, but sometimes the 'slots' available for a profile aren't cleared even though a dialog has finished. This seems to happen when an high amount of regular traffic is hitting kamailio. Has anyone else experienced this? Are there perhaps any other ways to control the slots available for certain incoming SIP traffic? Below parts of my configuration file. Perhaps I'm doing something wrong which causes this to happen. In my route[WITHINDLG] -> loose_route -> is_method("BYE") I'm not calling dlg_manage(), could this be the problem? Other comments are welcome too, as I'm fairly new to Kamailio. modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value", "318005004"); modparam("dialog", "default_timeout", 43200) modparam("dialog", "dlg_match_mode",2) modparam("dialog", "detect_spirals", 1) request_route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE")) { xlog("L_INFO", "Incoming request for $rU from $fU"); if ($rU == "318005004") { route(DIALOG); } } # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[DIALOG] { dlg_manage(); record_route(); $var(SIZE) = 0; sql_query("vf", "exec dbo.SelectAvailableSlots '00$rU'", "ra"); $avp(s:limit) = $dbr(ra=>[0,7]); xlog("L_INFO", "Maximum simultaneous calls is configured at $avp(s:limit) ...."); sql_result_free("ra"); if ($rU == "318005004") { get_profile_size("318005004", "$rU", "$var(SIZE)"); } xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, excluding the current call"); if($var(SIZE) >= $avp(s:limit)) { xlog("L_INFO", "Simultaneous calls limit of $avp(s:limit) reached for $rU: $var(SIZE)\n"); sl_send_reply("603", "Simultaneous calls limit reached"); exit; } if ($rU == "318005004") { set_dlg_profile("318005004", "$rU"); } if ($rU == "318005004") { if(get_profile_size("318005004", "$var(SIZE)")) { xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, including the current call"); } } } route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #Used for call transfer if($fU == "318005004") { if (is_method("UPDATE")) { sl_send_reply("405","Method not allowed"); xlog("L_INFO", "Dropping UPDATE message...\n"); exit; } if (is_method("INVITE")) { record_route(); } if (is_method("BYE")) { dlg_manage(); } } if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } Regards, Grant

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status of kamailio stun server?
by Juha Heinanen 16 Dec '12

16 Dec '12

i tried to use kamailio built-in stun server, but failed, because it supports only binding requests. my ua tried to send it "allocate request lifetime" request, which failed with 600 global failure response. what is the status of kamailio's stun server? is anyone interested in developing it further or should we obsolete the server and start using some external server? -- juha

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Writing User-preferences Username into Radacct Table on Call Forward
by Jonathan Hunter 15 Dec '12

15 Dec '12

Hi All, I was wondering if anyone would be kind enough to let me know if they have implemented anything like as follows in Kamailio? We are looking to implement call fowarding, activating it as usual in usr_preferences table. I was wondering if on the outbound leg its possible to write the username from usr_preferences with CFW active into the radacct table easily? This is with Radius and opensips DBs in Mysql. Many thanks Jon

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NAT Traversal :: RTPProxy and Jitsi client
by Raj Roy Ghandhi 14 Dec '12

14 Dec '12

Hi, I am trying to communicate 2 Jitsi clients in 2 separate private networks. Both clients are behind NAT. Kamailio Server is on Public IP. Both Jitsi clients does register and presence works fine. And also text messages works well. But unable to call each other. Had some one had the same issue before. I tried with many soft clients and all failed with same result. Same scenario works with IP Phones. Can anybody point out what I miss there. Regards, Roy.

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OT: MSRP over WebSocket Javascript stack
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi, Crocodile has just open-sourced our MSRP over WebSocket (see http://tools.ietf.org/html/draft-pd-msrp-websocket) Javascript stack. The project is hosted on Google Code: http://code.google.com/p/crocodile-msrp/ The stack is distributed using the MIT License and was developed and tested along side the Kamailio MSRP over WebSocket implementation. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

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RTP Proxy Interaction
by Austin Einter 14 Dec '12

14 Dec '12

Dear All I am looking at Kamailio's module, file, function that receives rtp and rtcp port from rtp proxy. Can somebody please let me know which file/function I need to check. Thanks Austin

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Re: [SR-Users] Help
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi Roy, Please keep emails on list. Regards, Peter On Fri, 2012-12-14 at 09:31 -0800, Raj Roy Ghandhi wrote: > Dear Peter, > > Thanks for all the support that you gave me during the Kamalio > configuration with RTPProxy. > Currently it works fine with IP Phones but not soft clients. (Jitsi) > > > If I give you my server information would you be able to check that > out for me. > > > I was lost for that issue. > > > Best Regards, > Roy. -- Peter Dunkley Technical Director Crocodile RCS Ltd

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Local call routing accross many kamailio servers
by Ali Jawad 14 Dec '12

14 Dec '12

Hi I do have 3 kamailio servers sharing the same db, calls go to Asterisk and that works great, I did purchase a DID and point it to one of the kamailio servers, it works fine so far for that kamailio server as the call is forwarded to the softphone ..but ONLY if the user is registered on the kamailio server the DID is forwarded to..if the kamailio user is on another kamailio server it does not work..although the database and location tables are shared. Simple diagram DID Provider <=== DID Kamailio ===> Kamailio Servers A, B, C If a softphone registers on DID Kamailio and I call the DID it works, if a softphone registers on Kamailio A or B or C, I never get the call, looking at ngrep the call just stays on DID Kamailio and it does not try to send it to any other server, how can I make the Kamailio DID lookup the destination from the location table and send the call there ? Regards

2 9

Hardware requirements voip servers
by Grant Bagdasarian 14 Dec '12

14 Dec '12

Hello, We are considering upgrading our VoIP platform and I'm wondering what kind of hardware we need for our requirements. The architecture will consist of a SIP Proxy Server (Kamailio) in front and a cluster of Media/Application Servers behind it. The requirements are based on a single server. SIP Proxy Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. - RTP Proxy - Record-Route Media/Application Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. What do we need in terms of CPU, Memory, HDD when looking at the following specific requirements: - 100 CPS - 200 CPS - 300 CPS Which hardware components are critical for handling this many calls, is it CPU, Memory, Disk IO bound? Also which open source media/application server software is suitable for these requirements? We are currently using Asterisk, which works great, but perhaps there are others available which are more suitable for these requirements. Thanks in advance! Regards, Grant Bagdasarian

1 0

Kamailio <-> Asterisk MWI
by Daniel Tryba 13 Dec '12

13 Dec '12

I'm trying to get the MWI indicator on endpoints to work by redirecting the SUBSCRIBE to the voicemail server (asterisk) using usc_replace_(to|from). This works fine for the initial request but the next subscribe will fail since the to/from don't get rewritten, which is indicated by the error: ERROR: uac [replace.c:250]: decline FROM replacing in sequential request in auto mode (has TO tag) This occurs when the modparam "uac","restore_mode" is manual or auto. When set to none the redirect will be correct but all responses have to be rewritten manually (I'm a beginner and I don't want to do this just yet). And since uac has knowledge about this having being rewritten, sending the original from/to looks like like a bug to me (more probable a misconfiguration/omission of something crucial withing dialog responses :) Changes to the (default debian package) config, in route route[PRESENCE] in if(is_method("SUBSCRIBE")) if( is_method("SUBSCRIBE")) { if(search_hf("Event", "message-summary", "a")) { if (!www_authenticate("$fd", "subscriber")) { www_challenge("$fd", "1"); exit; } if(avp_db_query("SELECT value FROM usr_preferences WHERE username='$au' and attribute='voicemail' ORDER BY value limit 1")) { $avp(voicemail)=$avp(i:1); $avp(subscrfrom)=$fu; avp_subst("$avp(subscrfrom)", "/sip:.*@/sip:$avp(voicemail)@/"); uac_replace_from("$avp(subscrfrom)"); uac_replace_to("","sip:$avp(voicemail)@$sel(cfg_get.voicemail.srv_ip): $sel(cfg_get.voicemail.srv_port)"); remove_hf("Authorization"); $ru = "sip:" + $avp(voicemail) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); } } else { handle_subscribe(); t_release(); } } exit; 192.168.34.226 is the sip device, 192.168.32.40 kamailio, 192.168.32.41 the voicemailserver. Initial request: U 192.168.34.226:5852 -> 192.168.32.40:5060 SUBSCRIBE sip:grandstream@sip.local SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>. Gets rewritten to: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:0880100782@sip.local>;tag=67d8f67065827fe2. To: <sip:0880100782@192.168.32.41:5060>. The response gets relayed to the endpoint and MWI will light up since there is a message. But next subscribe update from the endpoint triggers the "decline FROM replacing in sequential request" error and thus the subscribe send to voicemail is the original to/from: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>;tag=as324de8f3. Resulting in a 404 since the to/from are not the real peername on the voicemail server. How do I force Kamailio to rewrite the from/to without doing this manually for a requests/responses (restore_mode none)? The easy way out of this problem would be to have a 1 to 1 relation for the Kamailio subscribers and voicemail peers. But uac_replace should work (somehow). Attached is a full capture of the subscribe/notify. Running 3.3.2+squeeze from http://deb.kamailio.org/kamailio/. Machine is also running topoh module (192.168.0.1). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 6

NAT for Load Balancer and Proxies
by Nguyen Anh Tuan 13 Dec '12

13 Dec '12

Hi everyone, I have 1 Load Balancer and 2 SIP Servers, all of them are Kamailio. Does anyone suggest how to add NAT for them? Best Regard, Nguyen Anh Tuan

2 1

SIP proxy changing Contact header of presence server requests
by Owen Lynch 13 Dec '12

13 Dec '12

Hi, we have 2 separate instances of kamailio running as a SIP proxy and presence server. The proxy uses the nathelper module to correctly change the Contact headers of client requests. However it also does the same for notify requests from the presence server, which results in incorrect RURI from subsequent client subscribes. The NATDETECT route is shown below, I plan to leave the existing nat_uac_test but also explicitly skip the fix_nated...() calls for requests from the presence server. Does this seem reasonable? # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } Thanks, Owen Lynch

2 1

Dispatching to Asterisk cluster
by Grant Bagdasarian 12 Dec '12

12 Dec '12

Hello, We have a cluster of three Asterisk machines. Each machine answers an incoming call and transfers it to unique remote destination. So, Asterisk01 transfers to Destination01, Asterisk02 to Destination02, Asterisk03 to Destination03. The Asterisk machines are allowed to receive and answer only 50 calls each. I have a Kamailio in front of these three machine. It is currently used to dispatch calls to the three machines using round-robin. But I need some way to equally dispatch a maximum of 50 calls to each device in the Asterisk dispatcher set. I was thinking of using the dispatcher module and using the 10 algorithm, to load balance based on call load. But I'm not sure if that will guarantee a maximum of 50 calls being dispatched to each Asterisk. Theoretically, if I set the ds_hash_size to 7, which is 128 slots, at some point two of them should have 43 calls and one 42 calls. Also the value of ds_hash_size should be the power of two. Are decimal values allowed, like 7,229 which equals to almost 150? Perhaps I should use the ratelimit module for this? I'd appreciate it if someone could give me some advice on this. Regards, Grant

2 2

Syntax of xavp
by Andreas Granig 12 Dec '12

12 Dec '12

Hi, I'm playing around with xavp, but there are some things I can't wrap my head around. What basically works is this: $xavp(a=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; Getting the value like this: xlog("L_INFO", "a-foo='$xavp(a=>foo)' and a-bar='$xavp(a=>bar)'"); as well as xlog("L_INFO", "a-foo='$xavp(a[0]=>foo)' and a-bar='$xavp(a[0]=>bar)'"); ... works, because when omitting the index, it assumes [0]. I really like to avoid such implicit assumptions though, to make the config really clear, so I prefer the second approach. What doesn't work for me then is explicit assignment of the very first value, like this: $xavp(a[0]=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; If I do this, the "a" xavp is never created. Wouldn't it be good practice to create the xavp on first assignment? That way, I don't have to take care in the config file whether the xavp is used for the first time, where I have to omit the index. What I'm also wondering is whether it's possible to directly access nested xavps, like this: $xavp(a=>foo) = 'afoo'; $xavp(b=>foo) = 'bfoo'; $xavp(c=>a) = $xavp(a); $xavp(c=>b) = $xavp(b); xlog("L_INFO", "a-foo='$xavp(c=>a[0]=>foo)'"); or xlog("L_INFO", "a-foo='$xavp(c=>a=>foo)'"); Both ways give me "a-foo='<<xavp:0x7f0d387fe178>>'". Is this even intended? Would be really cool if that's possible! Thanks, Andreas

3 4

NAPTR, SRV and sips vs. transport=tls
by Andreas Granig 12 Dec '12

12 Dec '12

Hi, Hope to get some guidance here over the usage of "sips" and "sip plus transport=tls" when following RFC3263. The RFC3263 says that a NATPR record could return something like this for a query like "host -t NAPTR example.com": ; order pref flags service regexp replacement IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.example.com. IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.example.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.example.com. This means that the client should use sips if possible when contacting example.com. Concluding from that, I suppose it should perform an SRV lookup "host -t SRV _sips._tcp.example.com", and the result would be: ;; Priority Weight Port Target IN SRV 0 1 5061 server1.example.com IN SRV 0 2 5061 server2.example.com What I'm curious about is how requests towards one of these servers should look like, and how they are being handled by kamailio. A lot of clients and servers are sending "sip:user@example.com;transport=tls" in request URIs and Contact headers and Record-Route headers, and you can check with uri_param("transport","tls") which transport socket to use. This is pretty useful as you can determine hop-by-hop whether or not to use TLS. This approach has been obsoleted by RFC3261 though, and there doesn't seem to be a mechanism in RFC3263 to indicate "use schema sip, but use transport=tls". On encounter of a NAPTR record like the one above, how does kamailio act? Does it set a "sips" schema for the next hop? And what's the general take on this "sips" schema? As far as I understand RFC3261, it means that if a client sends a request to a sips-URI, the request is sent to the domain via TLS, and from there "the request is sent securely to the callee, but with security mechanisms that depend on the policy of the domain of the callee." (RFC3261, Chapter 4). What does this really mean in practice? Are you allowed to rewrite the schema to "sip" and pass it on for example via UDP to the callee if the callee didn't indicate transport=tls (deprecated anyways) or "sips:" in the Contact of the registration? Or should you keep "sips" as schema, but still send it via UDP, because you know based on local policy or based on client registration that the next hop is not supporting TLS? How would widespread clients react when getting a call to a "sips" URI, especially if they receive it via UDP? Looking forward hearing your input on that, Andreas

5 14

MySql error
by Hermes F. 11 Dec '12

11 Dec '12

I'm trying to create MySql database using the following /usr/local/sbin/kamdbctl create I followed the instructions on http://kb.asipto.com/kamailio:install:3.3.x-from-git-centos5x but I receive the following error message: "ERROR: database engine not specified, please setup one in the config script" Can someone help me please ?

2 1

Regarding Kamailio Registrar
by AMIT SHAH 11 Dec '12

11 Dec '12

Dear all, We are using Kamailio SIP Server in our application. We want to know 1 thing: We are using Kamailio SIP server as registrar and proxy server. For registration database, we are using dbtext module. We maintain record of SIP Extensions who have registered in Our proprietary file too. Now when we boot the Kamailio SIP server, At that time our file should be empty i.e. previously registered Extension entries should be deleted or it must be kept as it is. But if during power off, any extension had sent REGISTER request to Unregister then it was not executed as kamailio SIP server was not ON. But at boot time, as its entry was present, Kamailio SIP server will Consider it a registered extension. Please guide us for this situation. Thank you, Amit Shah.

2 1

Unable to set custom RPID
by Mino Haluz 11 Dec '12

11 Dec '12

Hi, what things should I set in order to set my custom RPID. What I am doing: remove_hf("Remote-Party-ID"); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); but RPID is only removed, second command does not append any rpid. Thanks Mino

2 1

New developer: Konstantin Mosesov
by Daniel-Constantin Mierla 10 Dec '12

10 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Konstantin Mosesov - he has contributed patches to app_python and joins the team to help maintaining and developing this module. His git commit id is: ez My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

How to use LIMIT 1 in db_query api
by Krishna Kurapati 10 Dec '12

10 Dec '12

Hi, I need a way to limit the results to just one row. The only way I know I could do is to call db_query followed by fetch_result of 1 row. However this is not optimal as all the rows are fetched to dynamic memory with db_query. Is there an alternative approach? Thanks Krish Kura

2 1

Help
by ameneh pourmoghadas 07 Dec '12

07 Dec '12

Dear Admin, My name is Ameneh from India. I am new in SIP. I have installed asterisk 1.8 on my ubuntu system and could have registered the internal users in my LAN to talk over x-lite. Now my question is how can I make a call to a mobile in India or a PSTN number in another country that would require a public SIP I think. how can I make my SIP server as public? would you please help me? thanks and regards, Ameneh

7 6

CTL module and alternatives for remote execution mi commands.
by Pavel Zalunin 07 Dec '12

07 Dec '12

Hi. In our project we need to query current users, dialogs and ability to end dialog using some rpc. Requirements are - do this remotely. I already tried to use xmlrpc module. When we had 42 users online it returned 17kb broken xml file. Then I tried to use binrpc protocol from CTL module. Usage ends with returning broken message too (in case, when message is biger than 21kb). Are there any other more stable solutions to query mi information using some tcp protocol? P.S. When I discovered sources of CTL module I found usage of writev - here is the problem about partial packets, when buffer is longer, than 21kb it don't pushes all vector buffers into socket. And in general, this module is using async sockets, AFAIK writev don't gives any guarantees, that it will write all vectors in async sockets. P.P.S. Of couse, last resort solution will be writing proxy for using mi_fifo on kamailio side to forward all records to other servers, but if there are more pretty solutions, please give me know:) Thanks. Pavel.

2 2

OFFTOPIC - Realtime testers needed - Asterisk Path header support (Oolong branch)
by Olle E. Johansson 07 Dec '12

07 Dec '12

While this is not Kamailio - it's related. If you want to use Kamailio in front of Asterisk, you do need SIP Path header support in Asterisk to do it right. Especially if you have one Kamailio handling Internet-facing communication and one handling internal communication. Please test! /O Vidarebefordrat brev: > Från: "Olle E. Johansson" <oej(a)edvina.net> > Ämne: Realtime testers needed - Path header support (Oolong branch) > Datum: 7 december 2012 11:07:09 CET > Till: Asterisk Developers Mailing List <asterisk-dev(a)lists.digium.com> > Kopia: "Olle E. Johansson" <oej(a)edvina.net> > > Friends, > > I have taken Klaus Darillion's patch for Path header support in Asterisk and created branches for 1.4, 1.8 and trunk. I need help testing it with realtime support by someone who are more used to the realtime databases than I am. I also need help completing the LDAP schemas with the new attribute. > > General testing is also encouraged. > > I have been using this patch without realtime for a couple of months. So far no Squirrels where hurt in the progress. And no bugs hit me in the back. > > Read more about the patch here: > http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/R… > > Klaus' original review: > https://reviewboard.asterisk.org/r/991/ > > The SIP Path header support is documented in RFC 3327 (that has a strange title). > > Basically, this is used when you have a proxy between a phone and your Asterisk. Asterisk needs to remember the path to the device when sending an INVITE, so we save both the contact and the path (basically a route) to the phone when Asterisk receives a REGISTER. When sending an INVITE (or anything based on the registry) a route header is added and the message finds the proper way to the device. > > Path header support exists in the path module in Kamailio if you want to test. > http://kamailio.org/docs/modules/3.3.x/modules_k/path.html > > Looking forward to your feedback! > > /O

1 0

Kamailio\ Message 300 Contacts
by Fatima Chahrour~Vanrise Support 06 Dec '12

06 Dec '12

Dears, In the redirect message 300, is there any way to control the maximum number of contacts(IP's) returned in the message? (am only getting 5 IP's) Also, if I have one gateway with 5 IP's and trying the same call 3 times; the order of the IP's returned in the message 300 is different in each time. Why is that? Thanks in advance, F Chahrour

2 2

Kamailio 3.X.X + BLF configuration trouble
by Hemanshu Patel 06 Dec '12

06 Dec '12

Hello everyone, Till now we were working with asterisk to handle BLF. But then we decided to move everything to kamailio and hence i started working on BLF with kamailio. I have already configured kamailio for Call handling and presence information. >From different pages on Internet, I have found some information about configuration of kamailio+blf. And i did so, but somehow things are not working as they should. I have done following changes to kamailio.cfg file loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_mwi.so" loadmodule "dialog.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" modparam("presence", "fallback2db", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") modparam("dialog", "db_mode", 1) modparam("pua", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") Now when my phone(yealink VP530), sends subscribe request it looks as below: SUBSCRIBE sip:1001@172.16.27.66 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK10464096 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66> Call-ID: 1572901975(a)172.16.27.61 CSeq: 1 SUBSCRIBE Contact: <sip:1000@172.16.27.61:5063> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: VP530P 23.70.0.40 Expires: 120 Event: dialog Content-Length: 0 In response to this message, Kamailio returns 202 with following information SIP/2.0 202 OK Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK245273058 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8ee2 Call-ID: 1572901975(a)172.16.27.61 CSeq: 2 SUBSCRIBE Expires: 120 Contact: <sip:172.16.27.66:5060> Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Followed by a NOTIFY NOTIFY sip:1000@172.16.27.61:5063 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.66;branch=z9hG4bK3ccd.adfbc416.0 To: sip:1000@172.16.27.66;tag=2058189864 From: sip:1001@172.16.27.66;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6cd8 CSeq: 1 NOTIFY Call-ID: 2040123403(a)172.16.27.61 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:172.16.27.66:5060> Subscription-State: active;expires=170 During subscribe message asked for XML based reply, yet NOTIFY doesnt send any XML. and hence there is no impact of this message on device. Nothing is changed.Also there's nothing when i call from this device to another, ideally i should get some NOTIFY messages So, i think i am doing something wrong in Configuration. Can anyone please help me in it. If you need any other info, i'll provide you. -- Regards, Hemanshu Patel Senior Software Engg P Help the environment – please don't print this email unless you really need to!

3 7

Internal/External SIP setup
by Zero Aggression 06 Dec '12

06 Dec '12

Hello, I am a bit new to VoIP, and I just need a quick yes/no (and maybe pointers) answer. Is the following possible (and smart) to do with Kamailio? Kamailio server. Dual-homed (internal network, Internet). Internal users connect to Kamailio to make calls to internal and Internet SIP peers. Users have NO internet connection themselves (they can only reach the server). Can Kamaillio (when dual-homed) transport all the data/voice between Internet and intranet? Can it also be used to carry all traffic between intranet users (if there is no direct route between them)? Thanks so much. Zane

3 3

db_cluster configuration
by Juan José Ivars 06 Dec '12

06 Dec '12

Hello, i am configuring db_cluster module in kamailio 3.3 I have configured in roun-robin read between two MySQL servers but the load is not distributed equally My configuration is that: modparam("db_cluster", "connection", "con1=>mysql://USER1:PASS1@1.1.1.1/DB") modparam("db_cluster", "connection", "con2=>mysql://USER2:PASS2@2.2.2.2/DB") modparam("db_cluster", "cluster", "cls1=>con1=9r9p;con2=9r9p") And the auth module with: modparam("auth_db", "db_url", "cluster://cls1") But i have enabled the mysql logs and the major part of the querys are targeted to the second server Is there any cache by peer? What am i doing wrong? Thanks in advance.

2 1

Preparing for the next major release
by Olle E. Johansson 06 Dec '12

06 Dec '12

Hi! Daniel has set a code freeze date for the next major release and the Kamailio community needs to work together to prepare the release. That means you as well! There will be a lot of work in the coming weeks to merge code from SIP router into the Kamailio core to complete the merger process. We also have a large set of new modules and functions to test. The creation of a new release is much more than code - it's testing, documentation proof-reading and feedback. With a large and complex piece of software like Kamailio we can't test all functions with all possible combinations of configurations and connections to third party products - this is where YOU can help out! Read more on http://www.kamailio.org/w/2012/12/preparations-for-new-release/ and join us in creating the new release! /O

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 06 Dec '12

06 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 06 Dec '12

06 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Segmentation fault in sip-capture
by Owen Lynch 05 Dec '12

05 Dec '12

Hi, I am running Kamailio 3.3 as a Homer capture agent, the sip-capture module receives HEP packets from a sip-trace module in a remote kamailio instance. Periodically the capture agent dies. The log and core dump suggest the initial problem is an invalid HEP msg received, but kamailio crashes when trying to send an error reply. Can anyone shed any light on this? Thanks, Owen Lynch Back trace: Program terminated with signal 11, Segmentation fault. #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 3967 tcp_main.c: No such file or directory. in tcp_main.c Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.i686 keyutils-libs-1.4-4.el6.i686 krb5-libs-1.9-33.el6.i686 libcom_err-1.41.12-12.el6.i686 libselinux-2.0.94-5.3.el6.i686 mysql-libs-5.1.61-4.el6.i686 nss-softokn-freebl-3.12.9-11.el6.i686 openssl-1.0.0-20.el6_2.5.i686 zlib-1.2.3-27.el6.i686 (gdb) bt #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 #1 0x08136e78 in handle_tcpconn_ev (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4310 #2 handle_io (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4362 #3 0x0813ea3c in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x080a502b in main_loop () at main.c:1727 #6 0x080a6c00 in main (argc=11, argv=0xbfeafd14) at main.c:2546 Log: Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: : <core> [pass_fd.c:288]: ERROR: receive_fd: recvmsg on 10 failed: Connection reset by peer Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: <core> [tcp_main.c:2347]: BUG: tcp_send: failed to get fd(receive_fd): Connection reset by peer (104) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sl [../../forward.h:193]: msg_send: ERROR: tcp_send failed Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=704 a=3 n=exit Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sipcapture [sipcapture.c:682]: ERROR: sipcapture:hep_msg_received: not supported version or bad length: v:[10] l:[10] vs [40] Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:638]: status: <500> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:640]: reason: <Invalid CSeq> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 234, <received> = <yadayadayada>; state=6 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK2348.d1620f62.0>; state=16 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:785]: child process 19576 exited by a signal 11 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:788]: core was generated Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19575]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19574]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19573]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19572]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tls [tls_init.c:674]: tls module final tls destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:242]: shm_mem_destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:245]: destroying the shared memory lock Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [main.c:804]: terminating due to SIGCHLD

2 5

modules_s/eval - survey of usage
by Daniel-Constantin Mierla 05 Dec '12

05 Dec '12

Hello, is anyone using the modules_s/eval module? In the context of merging modules_s/* to modules/*, this one lacks the docbook xml files for readme. It has a text readme file, from where I understand it does script operations using a stack and polish notation expressions. Probably it was developed before the expressions support in config file language, at this moment all its operations can be done directly by the interpreter. I am considering moving the module to obsolete/ folder. If anyone is using it and wants to keep it in modules/*, then he/she has to convert the readme text into docbook format (which is not complex at all, just takes a bit of time, probably like 60 min or even less). If there is no answer in one week about usage, will be sent to obsolete/, from where it can be rescued in case of need (just to be clear for everybody that code is not lost). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

dns resolver issue (RFC3263)
by MÉSZÁROS Mihály 05 Dec '12

05 Dec '12

Hi All, I am experiencing an issue when i try to contact xy(a)cisco.com. I found that kamailio/sip-router can't resolve by default resolving way a TCP + SRV records from domain cisco.com. e.g. cisco.com misi@alma:~$ host -t NAPTR cisco.com cisco.com has no NAPTR record misi@alma:~$ host -t SRV _sip._udp.cisco.com Host _sip._udp.cisco.com not found: 3(NXDOMAIN) misi@alma:~$ host -t SRV _sip._tcp.cisco.com _sip._tcp.cisco.com has SRV record 1 0 5060 vcsgw.cisco.com. misi@alma:~$ host -t SRV _sips._tcp.cisco.com _sips._tcp.cisco.com has SRV record 1 0 5061 vcsgw.cisco.com. I can't call xy(a)cisco.com because it does not exists an NAPTR record in domain cisco.com, and furthermore no SRV with udp. But it exists sip+tcp record and also exists an secure sip SRV, so sips+tcp record! I read rfc3263 I find kamailio dns resolver is not working as it should according RFC3263. http://tools.ietf.org/html/rfc3263 _If no NAPTR records are found, the client constructs SRV queries for those transport protocols it supports, and does a query for each._ So kamailio should query all (udp, tcp, tls, sctp whatever) protocols. Can anyone help/guide me to create a fix to this issue? My plan is to create a patch to kamailio resolver, to correct and behave according RFC3263. Any help or guidance appreciated! Thanks, Misi

4 14

syslog message about invalid domain name
by Juha Heinanen 05 Dec '12

05 Dec '12

when i t_relay in-dialog bye: Request-Line: BYE sip:0x8eb94d0@[2002:c062:660a::1]:5050;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP [2002:C062:660A:0:0:0:0:1];branch=z9hG4bKde53.6aaed77ea908953bbb133e5021d74036.0 Via: SIP/2.0/UDP [::1]:5090;received=0:0:0:0:0:0:0:1;branch=z9hG4bKq~ouJaGs;rport=5090 From: <sip:jh@as.test.fi>;tag=5A29C828-50BEFE55000C7118-B4FF9B70 To: "" <sip:test@test.fi>;tag=92e26ffbfbc16e53 CSeq: 10 BYE Call-ID: 28d1e4872cc5f83e User-Agent: Sip Express Media Server (1.6.0 (x86/linux)) Max-Forwards: 69 Content-Length: 0 with $du set to sip:[2002:C062:660A:0:0:0:0:1]:59933;transport=tcp, i get to syslog: Dec 5 09:53:48 sip /usr/sbin/sip-proxy[1947]: INFO: <core> [resolve.c:729]: invalid domain name '[2002:C062:660A:0:0:0:0:1]' since everything is and works as expected, why is the info message produced and how can i get rid of it? -- juha

2 2

Xcap server chooice among openSER, mobicents and openXCAP
by zhi sun 05 Dec '12

05 Dec '12

hi guys, it is my first time to use XCAP, i want to implement XCAP server to support maybe huge amount of user accesses (as cloud solution), so far, i can find out three XCAP server solution: 1. openSER with XCAP Server Module http://kamailio.org/docs/modules/devel/modules_k/xcap_server.html 2. mobicents with XCAP component/server build-in http://code.google.com/p/mobicents/wiki/Welcome?tm=6 http://www.mobicents.org/diameter/index.html http://hudson.jboss.org/hudson/view/Mobicents/job/Mobicents-Books/lastSucce… 3. openXCAP http://www.ag-projects.com/projects-products-96/101-openxcap but i am not clear to choose with solution to best match my situation (high volumn, high scale, and quick response). - does openSER support the cloud solution with high volum, high scale? - is it easy to use/integrate the XCAP server independently from mobicents? - is there performance issue for openXCAP, because it is implemented using python? any suggestions would be appreciated! best regards, -iamsy

1 0

io_wait errors
by Juha Heinanen 05 Dec '12

05 Dec '12

any idea what could be the cause for lots of these kind syslog messages: Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 56 Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 56 in the hash(-1, 0, 0xb2f28adc) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 59 in the hash(-1, 0, 0xb725a9d8) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 58 on 59: Broken pipe (32) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0xb3024a48 (flags 0x4018), fd 58 -- juha

2 2

[Kamailio-Users] BYE message not relayed to caller
by Vikram Ragukumar 04 Dec '12

04 Dec '12

Hello, My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy. I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like : No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: <sip:IPaddr_proxy:7160;lr=on;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" <sip:1234@IPaddr_VoipSwitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0 The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up. Any suggestions on how i can get around this problem ? Thanks and Regards, Vikram.

3 9

openser_receiver
by Uri Shacked 04 Dec '12

04 Dec '12

Hi, I notice that under my /tmp folder there are openser_receiver filles as well (kamailio_fifo and kamailio_ctl are know....) Version is 3.3.2 latest branch. Any ideas? Thanks, Uri

1 0

Problem with Stateless Dispatcher Call Load Distribution (Algorithm 10)
by SIP Mailing-list 04 Dec '12

04 Dec '12

Hi guys, I wonder if someone could sanity check something for me. I currently have a simple load balancer set up using algorithm 10 (call load distribution) from the dispatcher module. I'm using Record-Routing so that I can clear down the call load record at the end of the call. Everything looked like it was working okay until I just spotted a problem that I think might be a bug. It's to do with ACKs to non-200 responses. In a normal call, the flow goes: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 200 OK --- ds_load_update() 4 <-- 200 OK 5 ACK --> 6 ACK --> Now between step 3 and 4 the proxy runs ds_load_update which according to the documentation "set internal state to confirmed for the call load" entry in the internal call state store. This means the proxy knows where to send the ACK at step 5 because it is safely in the call state store so you can use ds_select_dst again on the ACK to get it to the right place. Works fine! Now consider the non-200 case: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 603 Decline --- ds_load_unset() 4 <-- 603 Decline 5 ACK --> 6 ACK --> ??? Between step 3 and 4 the proxy is supposed to run ds_load_unset() which unconditionally removes the call from the call state store. Now when the ACK comes in at step 5, the proxy has no record of the call and so doesn't know where to send the ACK. This results in retried Declines and ACKs. Broken :-( If the proxy doesn't run ds_load_unset() on the 603 response, is there some kind of timer that will cause the call to be removed from the call state store on unconfirmed calls? I don't think I can run ds_load_unset on the ACK because there's nothing in the ACK that tells me it is because of a 603, rather than a 200. Edit: actually there is! The Route header is present on the 200 ACK, but * not* on the 3xx to 6xx ACK. Maybe I can use loose_route... Anyway, any thoughts would be appreciated! Cheers, Richard

2 4

default flavour set to kamailio
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, based on community interaction, development, package distributions and documentation maintenance in the recent years, the default flavour has been switched now to 'kamailio'. As written in my previous email, we started the last phase to complete the integration between kamailio-ser, aiming to get to a unique database schema and set of modules in maximum 6 months. Kamailio is the most spread and popular flavour, presented in many linux/unix distributions, being the one released and packaged periodically. So it will stay as main flavour, the one officially released by the project. Those that want can take the sources and build their own flavour, as the support in Makefile system is kept. Once the integration is finished, the flavour system will just set the application name, nothing else. From the open source project perspective, the resources will be allocated only on releasing under the main/default flavour (Kamailio). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Last phase of integration for kamailio - ser
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, we plan to start the last phase of integration between kamailio and ser. After about 4 years with same source code tree, there is still some overhead caused by several duplicated modules. Following the decision to go on with the most developed components and absorb the missing features from the duplicates, kamailio modules are those that will stay (move to modules/ eventually) and get updates when needed. We used same decision to keep the core and makefile system from ser, adding to it what was extra in kamailio. I present next a summary of what is planned to happen: 1) Modules from modules_s/ that will be simply moved to modules/ because there is no conflict at all: - avp, avp_db, db_ops, eval, mangler, print, print_lib 2) Modules from modules_s/ that will be renamed and moved modules/ because it is mainly only conflict of name, but not in the backend (e.g., database): - auth_db => auth_dbattrs - because it uses (attribute,value) style to store subscriber profiles, which can be a preferred alternative for many out there - ldap => db2_ldap - it is a DB APIv2 connector to ldap server - bdb => db2_bdb - it is a DB APIv2 connector to berkeley db (to be double-checked) - xlog => xprintf - it uses %spec to print log messages with all the code inside the module - the only reason not discarding it and stick only to modules_k/xlog for the moment is because it is used from other modules, otherwise modules_k/xlog has more features (e.g., syslog facility, color printing, a.s.o) 3) Modules from modules_s/ that will be removed, after absorbing some extra features: - usrloc/registrar => get the ability to store AVPs per contact (modules_k is more advanced in features, like path/gruu/partial outbound support, publish presence states, etc.) - several modules must be reviewed: acc*, radius*, permissions, pike, gflags, options, timer 4) Modules from modules_s/ that will be removed, by moving to obsolete/ directory - the rest of them from modules_s/, like dialog, xcap, cpl-c, presence_b2b, dispatcher, dbtext, domain, etc. The categories and their content are not nailed down, it is the purpose of this discussion to see if something was overlooked. We hope to end this process in max 6 months time - after that will be a unique database schema and a unique set of modules. Flavour system in the Makefile will stay just for letting people choose easier the name of binary, otherwise everything will be the same (e.g., what is compiled, installed, etc). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, Kamailio SIP Server v3.3.2 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.1. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0 or v3.3.1. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.2. For more details about version 3.3.2 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/10/kamailio-v3-3-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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sample SCA kamailio.cfg added
by Andrew Mortensen 04 Dec '12

04 Dec '12

I've added a basic working kamailio.cfg with Shared Call Appearances to the SCA docs. It should help clarify where and when it's necessary to have the sca module inspect traffic. Those of you simply wanting to test SCA can just copy and paste the configuration, updating the listen address as necessary. Until the sca module doc on kamailio.org gets updated, you can find the sample configuration in the git master branch, or view it through gitweb at the link below. Best, andrew <http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mo…>

1 0

kamailio vs.ser
by Ovidiu Sas 03 Dec '12

03 Dec '12

Hello all, By inspecting the source code, the only difference that I could see between kamailio and ser flavours is that kamailio has support for the "tm:local-request". Are there any constrains in having the tm:local-request present for ser flavour? Does it make sense to continue to build two flavours? My suggestion would be to focus one one single flavour and go forward with it for the next release. Comments? Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

health check
by Alex Solt 03 Dec '12

03 Dec '12

Hi, Is there a way to perform health check on kamailio application? AS

2 1

LCR Failover
by Daniel W. Graham 03 Dec '12

03 Dec '12

I am having issues with LCR failure route. I have two gateways specified in LCR table, if I change the address of the gateway with highest priority to a bogus IP, the failure route never seems to takes place and call is never routed to second gateway. Also, I have tested successful calls to both gateways by switching the priorities around. Here is what I receive after the first gateway times out: U kamailio:5060 -> asterisk:5060 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK633b44d8;rport=5060. From: "user" <sip:user@asterisk>;tag=as67c0037f. To: <sip:dialednumber@domain>;tag=d7177f97ed67076dddc3163c85ab0656-f2e5. Call-ID: 248d69ed6495d0a65315458e2ba6c528@asterisk:5060. CSeq: 102 INVITE. Server: SIP Proxy. Content-Length: 0. Config: modparam("tm", "fr_timer", 10000) modparam("tm", "fr_inv_timer", 120000) modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") route[LCR] { if (!load_gws(1)) { sl_send_reply("503", "Internal Server Error, Unable to load gateways"); exit; } else { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { sl_send_reply("503", "Service Unavailable, No available gateways"); exit; }; }; exit; } failure_route[LCR_FAIL] { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { t_reply("503", "Service Unavailable, No gateways"); exit; }; exit; }

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Kamailio World - Conference & Exhibition - April 16-17, 2013, Berlin, Germany
by Daniel-Constantin Mierla 03 Dec '12

03 Dec '12

Kamailio project is announcing its first dedicated event - Kamailio World - to take place in Berlin, Germany, during April 16-17, 2013. The event is organized by Asipto (http://www.asipto.com) in collaboration with Fraunhofer Fokus Institute (http://www.fokus.fraunhofer.de). The conference is targeting to present commercial products and services build around Kamailio and related projects, as well as technical workshops of how to use Kamailio for various purposes. Along with the conference will be space for exhibition, that will allow companies to make demos and show cases for participants. For more details, visit the web site of the event: http://conference.kamailio.com Besides Kamailio, you will have the chance to interact with other open source VoIP projects such Asterisk, FreeSwitch, SIP Express Media Server, Homer SIP Capture, Siremis, a.s.o., as well as with commercial vendors in real time communications market. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Jitsi to Jitsi call
by Raj Roy Ghandhi 03 Dec '12

03 Dec '12

Hi Peter, Still I am unable to make call to Jitsi to Jitsi. Call established But no media. But text message works. NAT traversal work with RTPProxy and I can do IPPhone to IPPhone call and voice works very well. Tried with many soft phones and voice did not go through. Did I miss something in the configuration ? Please advice me. Best Regards, Roy.

2 2

Call pickup through a NAT
by Mark Boyce 03 Dec '12

03 Dec '12

Hi All, I've hit my next stumbling block on my mission to get BLF/Call pickup working on Linksys/Cisco phones which are behind a NAT working with Kamailio. I'm using RFC 3265 and pua_dialoginfo to handle subscribes from phones and generate notify packets on status change. The phones are on two different sites behind a couple of NATs. Kamailio is on public IP in the data centre. BLFs work fine. However the <target uri="x.x.x.x."> given in the XML of the notify packet is the that of the Public IP of the NAT so that call pickup fails. I'm guessing that this actually needs to be the address of the Kamailio server? I cannot see where to correct this without breaking tables the server needs to know how to contact the phone when it needs to. Has anyone seen this before? Cheers Mark

2 1

Kamailio SCA Module
by Robert Boisvert 01 Dec '12

01 Dec '12

Andrew, Congratulations on your new position as a Kamailio developer! When I saw your module (SCA) I began testing immediately since it may be a stepping stone to help me get the Polycom barge-in feature working with Kamailio. However, I ran into a few difficulties. Can you assist me? To make your existing code work as a K module I had to change a few lines of code and recompile under modules_k. Specifically, since the K definition of the ucontact structure defines ucontact->aor as str* and the S definition as str**, I changed “&c->aor” to “c->aor” in the following lines of sca_usrloc_cb.c: 73: if ( !sca_uri_is_shared_appearance( sca, c->aor )) { 74: LM_DBG( "%.*s is not a shared appearance line", STR_FMT( c->aor )); 79: &SCA_EVENT_NAME_CALL_INFO, c->aor ) < 0 ) { Since I use PostgreSQL as my SQL engine I created the SCA entries using this code: INSERT INTO version (table_name, table_version) values ('sca_subscriptions', '0'); CREATE TABLE sca_subscriptions ( id SERIAL PRIMARY KEY NOT NULL, subscriber VARCHAR(255) NOT NULL, aor VARCHAR(255) NOT NULL, event INTEGER NOT NULL, expires INTEGER NOT NULL, state INTEGER NOT NULL, app_idx INTEGER NOT NULL, call_id VARCHAR(255) NOT NULL, from_tag VARCHAR(128) NOT NULL, to_tag VARCHAR(128) NOT NULL, notify_cseq INTEGER NOT NULL, subscribe_cseq INTEGER NOT NULL, CONSTRAINT sca_subscriptions_idx UNIQUE (subscriber, call_id, from_tag, to_tag)); I applied the recommendations from your documentation on configuring Kamailio (3.3.1) and inserted the call to “sca_handle_subscribe ()” in kamailio.cfg. Next, I connected two Polycom phones (SP335 and SP650) to my network and configured them to enable presence and use a shared line. Four lines were configured on the phones as shown below: SP335 Line 1 (p11) = private line 6013 SP335 Line 1 (p12) = shared line 1519 SP650 Line 1 (p21) = private line 6014 SP650 Line 2 (p22) = shared line 1519 My tests get inconsistent results on the line LEDs and it seems that they do not work as a shared line. When I ran a tcpdump of the SIP communication (see attached pcap files) I found several anomalies: - Some packets (sca-reboot1 #22, 44, 113, 120 and sca-reboot2 #17, 38) contain spurious NUL characters in the tags section. In all tests it seems to occur in the Expire tag when Kamailio accepts the subscription. - When Kamailio sends call-info NOTIFY the Content-Length tag is missing. - Sometimes the phone does not respond to a call-info NOTIFY event ( sca-reboot1 #23, 45 and sca-reboot2 #18, 39). - I don't have an example here but sometimes the phones respond to the call-info NOTIFY with "481 Call Leg/Transaction Does Not Exist". I think some of my problems are related to these anomalies. Could you help me to determine if this is a problem with the code itself, my failure to properly port it to the K module, or some other issue? Once again, thank you for making this very useful module for Kamailio and for the documentation to make it work. I hope that resolving my problems may help the code to become a part of Kamailio. Sincerely yours, Bob Boisvert

2 1

Linphone presence
by Moacir Ferreira 30 Nov '12

30 Nov '12

I am trying to get Linphone to show presence information. I did my Kamailio setup using just the template from a stardard install and it is working properly with Bria (X-Ten). However, when using Linphone it does not show buddies presence. Any hint? Cheers! Moacir

2 1

Call forking
by Moacir Ferreira 30 Nov '12

30 Nov '12

All the topics/examples I found on the net is not what I am looking for, so here is the question: How do I achieve parallel forking (call all devices) if I have a single user registered using the same account from different devices (i.e.: table phone, iPad and PC)? What I want is to ring all devices a user may be using. Cheers! Moacir

2 1

t_fork_to?
by Juha Heinanen 30 Nov '12

30 Nov '12

core cookbook reads: append_branch Similarly to t_fork_to, it extends destination set by a new entry. The difference is that current URI is taken as new entry. what is t_fork_to? i could not find it in tm README. -- juha

3 3

WITH_PRESENCE charsh Kamailio
by Raj Roy Ghandhi 30 Nov '12

30 Nov '12

Hi, I am running Kamailio 3.4 with web-socket and everything works fine with NAT traversal for IP Phones. But when I enable the PRESENCE it crash with "(3) errors in config file" I have copied the config.cfg to paste-bin at http://pastebin.ca/2257817<http://pastebin.ca/2257815> Please guide me on this issue. Best Regards, Roy.

2 1

Enforced TLS, still some packets are through UDP
by Kamal Palei 30 Nov '12

30 Nov '12

Dear All I ma trying to have all SIP signalling through TLS. Using Kamailio 3.1.5. In a typical setup, endpoints initiates call, messages reach Kamailio proxy, Kamailio proxy forwards these packets to main proxy. In Kamailio.cfg, I have added below code. route { # per request initial checks route(REQINIT); setflag(FLT_NATS); if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.19:443"); exit(); } else { xlog("incoming request\n"); route(WITHINDLG); route(RTPPROXY); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); t_relay_to("tls:115.114.48.19:443"); exit(); } .............. I am finding most of the messages are getting transmitted in TLS. But few ACK and REFER / REFER response packets are still being transported in UDP. Can somebody tell me whats the wrong I am doing here. By any chance route(WITHINDLG); route(RTPPROXY); before t_relay_to("tls:115.114.48.19:443"); can cause this ???? And one more doubt.., 115.114.48.19 I get in route header. How can I extract the value of route header, so that I do not have to hard-code Many thanks in advance. -kamal

3 2

Lua env not initialized in event_route
by Vicente Hernando 30 Nov '12

30 Nov '12

Hello I'm trying to execute a Lua script on kamailio startup. I'm calling the script inside event_route[hable:mod-init], but it fails to be executed with this error: [app_lua_mod.c:167]: Lua env not initialized It happens both with lua_run and lua_dofile. I've played with the order in wich the modules are loaded, but I haven't been able to get rid of that message. Kamailio version is 3.3.0. Do you think this is feasible or should I try a different solution? Thanks in advance Regards, Vicente.

2 1

Presence on Cisco/Linksys handsets
by Mark Boyce 29 Nov '12

29 Nov '12

Hi All, I've been trying to get some Cisco/Linksys SPA phones working with Kamailio (current stable). All seems to be ok apart from getting Presence/BLF/SCA working. The phones are set to Server_Type : RFC3265_4235 Line Key 2 Extension : Disabled Line Key 2 Share Call Appearance : Private Line Key 2 Extended Function : fnc=blf+sd+cp;sub=1001@ourserver.com;ext=1001@ourserver.com Where 1001 is the other phone. The phone appears to be sending event type 'call-info' rather than dialog etc. Phone is being sent '489 Bad Event' and the error in the logs is : presence [subscribe.c:1007]: Unsupported event header field value call-info Whatever I set the Server Type to on the phones it still tries to send call-info events on subscribes Has anyone got Presence working with Cisco SPA handset? The module needed seems to be available on OpenSIPs http://www.opensips.org/html/docs/modules/1.8.x/presence_callinfo.html Thanks! Mark

3 8

Kamailio getting started
by Scott, Matt 29 Nov '12

29 Nov '12

Anybody got a good tutorial or starter project for Kamailio? We have Siremis 3 installed, but there is 0 help on that? Trying to solve 2 scenarios: -asterisk to softphone through kamailio -softphone to asterisk through kamailio. I ultimately want to use Kamailio as an edge router Any help is appreciated. Matt Scott

3 3

Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 29 Nov '12

29 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I’ve setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. ‘kamctl dispatcher show’ shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

2 13

Call from SipML5 -> PSTN
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, >From all of your support now I can call from 1. IP Phone <--> IP Phone 2. Web Page <--> Web Page 3. IP Phone -> PSTN without any issue But when I try to call from Web Page to PSTN then it tries to call sip:00xxxxxx89078@mysipdomain.com and that time out. Trying to figure out how to get this work ? Can anybody guide me on this please. Best Regards, Roy.

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Re: [SR-Users] CRLF from Kamailio server
by Klaus Darilion 28 Nov '12

28 Nov '12

forwarded to mailing-list .... -------- Original Message -------- Subject: Re: [SR-Users] CRLF from Kamailio server Date: Mon, 26 Nov 2012 16:34:26 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: miconda(a)gmail.com To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Hello, I haven't checked actually the parts for tcp, but recently I added the feature of removing the location records if there is no reply to OPTIONS keepalives in nathelper, and it was only for UDP. Also, I think the nat ping is 4 bytes (zeros) or so. Might be another option from tcp, like: http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_keepalive Cheers, Daniel On 11/26/12 3:54 PM, Klaus Darilion wrote: > IIRC tcp_crlf_ping is only about the Ping-Pong mechanism: client sends > PING=CRLF, server responds with PONG=CRLFCRLF. > > Thus, server should only response on client CRLF. Further the PONG is > always double-CRLF. > > So I suspected it is NAT keepalive - maybe someone added > NAT-keep-alive for TCP :-) > > regards > Klaus > > On 26.11.2012 11:05, Daniel-Constantin Mierla wrote: >> Hello, >> >> this might be actually related to tcp connection keepalive: >> >> http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_crlf_ping >> >> IIRC, nathelper pings only udp contacts. >> >> Cheers, >> Daniel >> >> On 11/26/12 10:54 AM, Klaus Darilion wrote: >>> >>> >>> On 26.11.2012 05:13, Dmytro Bogovych wrote: >>>> Greetings. >>>> May anyone give me advice? >>>> I have working Kamailio with TLS. >>>> It is used by iOS softphone - i developed own one based on >>>> resiprocate. >>>> All signalling is TLS. >>>> Almost everything is fine. >>>> >>>> But looks kamailio sends CRLF messages sometimes. >>>> If it happens when softphone is in background mode - it is killed >>>> by iOS >>>> (as iOS prohibits too often awakes from background mode). >>>> >>>> How I can disable these messages? >>>> I see natping_interval = 30 and ping_nated_only=0 >>>> Should I set natping_interval to zero ? >>> >>> Either set the interval to 0 or set ping_nated_only=1 and make sure to >>> not set the NAT flag for the iphone clients. >>> >>> A problem may be firewalls/NAT between the iPhone and Kamailio, as >>> they may drop the TCP connection is there is no activity for too long >>> time. >>> >>> regards >>> Klaus >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Call from SipML5 -> PSTN
by Peter Dunkley 28 Nov '12

28 Nov '12

You need some sort of media gateway or server. I don't know of any that currently support this. Peter Raj Roy Ghandhi <roy.gandhi(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

memcached loading problem
by Alex Balashov 28 Nov '12

28 Nov '12

Hello, So, I built the memcached module against the latest Git production branch of memcached and libmemcache-1.4.0.rc2 (the latest available), on Debian i386. But when I try to load the Kamailio memcached module: Nov 28 03:47:44 diminuendo-1 kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/memcached.so>: /usr/lib/libmemcache.so.0: undefined symbol: mcm_buf_len I find that very odd, because the symbol is quite clearly exported in the library: diminuendo-1:~# nm /usr/lib/libmemcache.so.0 | grep -i mcm_buf_len U mcm_buf_len Does anyone have any idea what the underlying linking issue is here? -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 4

Websocket with Kamailio 3.2.2 release
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, I am trying to integrate the websocket module into release 3.2.2. But I am unable to do that. Please guide me to get it done. Regards, Roy.

6 17

Eavesdropping SRTP sessions
by Mino Haluz 28 Nov '12

28 Nov '12

Hi, maybe it is not that kamailio related question, but I dont know any other place with such good voip professionals ;) I have kamailio and mediaproxy. Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am forcing clients to use SRTP but it does not support adding any certificate on both sides. SRTP call is working fine. The question is, in this case, is man-in-the-middle attack possible? Maybe I should study SRTP more, but basically, if there are no certificates, there is no method how to be 100% sure that the media goes directly between clients. Is it true? Thanks for response, Mino

2 4

Managing expiration values with memcached
by Alex Balashov 28 Nov '12

28 Nov '12

According to the memcached module docs, setting $mctex(key) can be used to modify the expiration value of a key. Should this be done after or before setting the key itself with $mct(...)? I have tried it before and the assignment didn't work, so clearly the key must first exist in memcache: $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:354]: setting pvar failed Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:407]: assignment failed at pos: (164,37-164,69) OK, so that makes sense. I tried it the other way: $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); But when I look on my memcached server, the expiration value is not being modified. It's still the default 'expire' value set in the memcached modparam. I ran the memcached server with maximum verbosity and verified that no TOUCH command is taking place, and that the SET command for the key contains the default expiration value, not the one I set with $mctex(...). So, it seems to me that the module is not committing this new expiration value when it is written to $mctex(...). Is this a bug, or am I missing something? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 2

problem with kamailio and libradiusclient-ng/freeradius-xs
by Rabary 27 Nov '12

27 Nov '12

Hi all, When I have this line in kamailio.cfg: modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") Kamailio failed to start: kamailio error, failed to start (/var/run/kamailio/kamailio.pid exists). I already try to stop kamailio, an remove this file but don't work In log,I have: Nov 27 20:13:39 voip-core-1 kernel: [14318.988658] kamailio[8370]: segfault at 0 ip 00007f0b0b94ca5d sp 00007fffa08f9460 error 4 in libradiusclient-ng.so.2.0.0[7f0b0b946000+c000] Nov 27 20:13:39 voip-core-1 kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe When I remove just this line and restart kamailio, it works, I don't know what is the problem because before I can do accounting with freeradius with the same config I have kamailio 3.3.2, kamailio-radius-modules 3.3.2,libradiusclient-ng2 0.5.6-1.1 and on 2nd server freeradius-xs and freeradius-xs-mysql 1.1.7-4 I put in attached file a few config of kamailio and radiusclient Regards -- Rabary Teddy Inutile d'imprimer ce mail

1 0

Disconnection from Kamailio
by Dmytro Bogovych 27 Nov '12

27 Nov '12

Greetings. I observe strange behavior with Kamailio. My iphone 3gs run softphone based on resiprocate. First 10-20 minutes everything works fine. After it the connection is closed from server (select() loop returns read notification, read attempt returns zero bytes). Ok, softphone closes old socket and create new one by new registration. But this connection is terminated from server in 4-5 seconds after registration finish (200 OK). Softphone makes next attempt to register - connection is closed after registration again. So it is infinite loop. After several attempts iOS kills application as it awakes too often in background mode. I have not logs from server yet; but maybe anyone had similar issue? Thank you :)

1 0

send 100 provisioning delayed
by pars3c 27 Nov '12

27 Nov '12

Hi , The proxy send the "automatic" provisioning message 100 "Trying" , near 1-2 seconds later the INVITE message, so the client send 3 times the same invite (correct , 500 ms timer). Is there any parameter to set , for the time between the INVITE and "100" message ? Is correct to disable the automatic send, and set after the invite, sl_send_reply("100", "Trying"); ? Thanks

2 1

Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 27 Nov '12

27 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I've setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. 'kamctl dispatcher show' shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

1 0

Kamailio crash
by Ricardo Martinez 27 Nov '12

27 Nov '12

Hello In the last three days my kamailio process has crashed two times. I obtained the core file and see it, this is the output. Can someone check what is the problem? Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 56 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 #1 process_mi_params (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1057 #2 mi_terminate_dlgs (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1165 #3 0x00007fc789996370 in run_mi_cmd (rx_sock=11, tx_sock=12) at ../../lib/kmi/mi.h:77 #4 mi_datagram_server (rx_sock=11, tx_sock=12) at datagram_fnc.c:525 #5 0x00007fc789998509 in datagram_process (rank=<value optimized out>) at mi_datagram.c:346 #6 0x00007fc78999a09c in mi_child_init (rank=<value optimized out>) at mi_datagram.c:296 #7 0x00000000004dbd61 in ?? () #8 0x00007fc78d99aab8 in ?? () #9 0x00007fc78d0181cd in pkg_proc_update_real_used (data=<value optimized out>) at pkg_stats.c:141 #10 0x000000000052b423 in ?? () #11 0x00007fc78d8a2830 in ?? () #12 0x0000000000000000 in ?? () (gdb) I’m using kamailio [root@pxh ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 2a00c6-dirty with dialog module, hash tables, radius authorization. Hope someone could help me. Regards, Ricardo.-

2 8

plans for next major release
by Daniel-Constantin Mierla 27 Nov '12

27 Nov '12

Hello, following the devel IRC meeting from last week, I am starting the discussion about the next major release planning. There were two relevant options: 1) freeze by end of January (2013) and release by March 2) freeze by end of December (2012) and release by February Many devels announced quite some work ahead, so the 1st option seemed not to be enough for finishing intended work, which may add some extra delays (several estimated to finish by end of February or March). So the 2nd came into debate, to do an early release as many people seemed very interested in getting websockets support out in a stable release. Both of the options fit in our releasing policy (8 to 10 months), first one with potential to delays. We do have now lot of new stuff in devel branch, with other ongoing work planned to be pushed before end of year. A suggestion was to do the following major release a bit shorter, if it proves to collect a lot of additions quickly -- maybe by summer 2013. As not all devels were in the meeting, but also because it impacts the user community, I am looking to get some feedback on the mailing lists as well. That will allow to make the decision quicker based on showed interest. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

One active registration per sip instance.
by Krishna Kurapati 26 Nov '12

26 Nov '12

Hi, I have 3.x Kamailio with GRUU support. I see SIP Instance in the contact. However, I see a problem. When I kill the SIP client on a device and restart, it sends registration on a different port. The server instead of updating the contact in user location table, is creating a new contact for the same instance. Do I need to turn on something to get the expected behavior? Here is the output from kamctl ul show. Thanks Krish Kura AOR:: 100227074 Contact:: sip:100227074@10.132.60.70:50845;transport=TCP;ob Q= Expires:: 3132 Callid:: ATZJcr.HqXxwWdsPmanjERetcdeRVeZy Cseq:: 4399 User-agent:: n/a Received:: sip:166.137.120.37:48873;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-50e-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50839;transport=TCP;ob Q= Expires:: 3029 Callid:: oYsIYUyP3N77MKSoQ3EDChEph.lD4fjB Cseq:: 37489 User-agent:: n/a Received:: sip:166.137.120.37:61763;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-507-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50803;transport=TCP;ob Q= Expires:: 1527 Callid:: b4IeJXybn6Yr.jB8sc-Kz8CJ0kt0hNbO Cseq:: 30014 User-agent:: n/a Received:: sip:166.137.120.37:36879;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5c-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50818;transport=TCP;ob Q= Expires:: 2199 Callid:: DM0Z1d-ZenGHknjmT.7Cle5cos7Uhs7H Cseq:: 22942 User-agent:: n/a Received:: sip:166.137.120.37:39471;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f6f-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50815;transport=TCP;ob Q= Expires:: 1833 Callid:: VJy3SNE66U1U971u5BEeqbYJi6a-puT- Cseq:: 64347 User-agent:: n/a Received:: sip:166.137.120.37:35872;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5a-2 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1

2 4

new developer: Andrew Mortensen
by Daniel-Constantin Mierla 26 Nov '12

26 Nov '12

Hello, I want to announce that a new developer got GIT write access to repository: Andrew Mortensen from University of Pennsylvania, USA - he has contributed a new module: sca (Shared Call Appearances for Broadsoft extensions). The module is right now in a personal branch, soon to me merged in master branch. His git commit id is: admorten My warm welcome and looking forward to future work within the project. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

ul.dump Problem
by Spencer Thomason 26 Nov '12

26 Nov '12

Hello, I'm attempting to dump the contents of the usrloc cache and I'm receiving the following error: sercmd> ul.dump error: 500 - Internal error creating aor struct sercmd> mi ul_dump error: 500 - Internal server error processing 's': buffer too small (overflow) (-2) It seems to work fine when there are only a small number of registrations (about 20 or so) but then fails when the number increases. It there a setting I need to increase somewhere? Thanks, Spencer

2 4

problem with $time Pseudo-Variables on 3.3.2
by Uri Shacked 26 Nov '12

26 Nov '12

Hi, I upgraded to 3.3.2 On 3.2 i used the variable $time() to get the current time for some process. Every 2 minutes a RTIMER process started, checked the $time(hour) and $time(min) and on a certain time executed something. This process stopped working on 3.3.2 When i debug, i see that the $time() variable keeps on giving me the same hour and minute that was given on the first run. Meaning the $time() variable does not give the current time. I notice also that the $Ts that on 3.2 gave a current time stopped doing it and $TS does it now.... Any help here? Is it OK and i need to check time another way? Or something in the $time() has changed...? Thanks, Uri

2 6

CRLF from Kamailio server
by Dmytro Bogovych 26 Nov '12

26 Nov '12

Greetings. May anyone give me advice? I have working Kamailio with TLS. It is used by iOS softphone - i developed own one based on resiprocate. All signalling is TLS. Almost everything is fine. But looks kamailio sends CRLF messages sometimes. If it happens when softphone is in background mode - it is killed by iOS (as iOS prohibits too often awakes from background mode). How I can disable these messages? I see natping_interval = 30 and ping_nated_only=0 Should I set natping_interval to zero ? Thank you :)

3 4

Re: [SR-Users] Forwarding Relaying of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi All Alex responded to my question saying the following: "You can relay SIP messages of any type just the same way as you do with INVITEs -- e.g. t_relay(), forward(). But, proxying registration messages can have complications." can anyone give me some pointers or a list of some of these complications so that I can further research to see how its is done. I will attempt using the t_relay to view the issues, just need some idea more specifically what to look for! Thanks

2 1

Re: [SR-Users] Re-Directing of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi Can anyone point to a good example of Kamailio performing the function of Re-direct server. We would like to forward messages onto another domain for authentication.

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