- sr-users - kamailio.org

[Kamailio-Users] pick_branch failed (lowest==-1) for code 302--HELP!!
by Krunal Patel 18 Aug '08

18 Aug '08

Hi, Here is the scenario: -----------------------Invite--------------> <----------100 Trying---------------------- <----------302 Moved temporarily-------- -----------------------ACK--------------> Now I would like to route call to contact returned in 302 responce. But I am getting problem: Aug 17 18:10:41 [16667] DBG:core:parse_headers: flags=80 Aug 17 18:10:41 [16667] DBG:tm:t_should_relay_response: T_code=100, new_code=302 Aug 17 18:10:41 [16667] DBG:tm:t_pick_branch: picked branch -1, code 999 Aug 17 18:10:41 [16667] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 302 Aug 17 18:10:41 [16667] DBG:tm:relay_reply: branch=0, save=0, relay=-1 Please help me to fix this. Thanking you Krunal Patel

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[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 17 Aug '08

17 Aug '08

I will be out of the office starting 08/15/2008 and will not return until 08/18/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

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[Serusers] Acc module extra patches
by caio 16 Aug '08

16 Aug '08

Hello, Is there patches for acc extra params for ser version 0.9.7? I've found this link http://www.voice-system.ro/docs/acc-headers/ but do not know if is it for v0.9.x Thanks... -- caio

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[Serusers] on Ubuntu Installation & Config
by Ameed Jamous 16 Aug '08

16 Aug '08

Hi all, Am trying to install & Configure SER on ubuntu 8.04. After i intsalled the SER file i can see this in my terminal: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.104 [192.168.1.104]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.104 [192.168.1.104]:5060 Aliases: tcp: Ubuntu-AM.local:5060 tcp: localhost:5060 udp: Ubuntu-AM.local:5060 udp: localhost:5060 ok now how can i add a user to register to SER ? How can i monitor the logs and the prosses happening? i tried using serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config Even though i have line fifo=/tmp/ser_fifo in my config. Can anyone help with this? Regards, Ameed -- Best Regards, Ameed Jamous Project Manager/Carriers Relations. Direct Line: +962 777 835819. www.joodtelecomgroup.com

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[Serusers] call forward no answer
by caio 15 Aug '08

15 Aug '08

Hi all, I can't make call forwarding on no answer works on my ser-0.9.7 installation. Here you can see a snippet of log, where the call falls on failure_route and then to route_1 where t_relay is called. Log at: http://rafb.net/p/xwTmn195.html The call is from sip:579951 to sip:579915(who has enabled the feature fwdnoanswer to sip:579916. But I do not know why the INVITE after the append_branch() is still to R-URI: sip:579915@.... After the relay then the call terminates with: Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ I follow the example of ser guide.., but do not work.. I would appreciate any comment.. Regards, -- caiogf

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[Kamailio-Users] Question
by Edgar Guadamuz 15 Aug '08

15 Aug '08

Hello, I'm a newbie and I don't know the difference between opensips (where openser.org redirects) and kamailio, If someone can give me a light on this it would be great, thanks,

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[Kamailio-Users] project foundation
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello everybody, I think it is the time to move forward with project management structure. During last days there were talks and discussions on the lists, the solution to make the project independent and long term trustable is to create a foundation. We made a draft of purpose and organization structure and we want to get your comments. http://www.kamailio.net/dokuwiki/doku.php/admin:foundation-draft For the statute we are looking at similar cases of foundations around open source projects. Considering the developer population and contributors, I find as good and reliable place to register the foundation to be Germany. There is still long way to go and a lot to discuss: * membership fees * how one can become a full member (associate member can be anyone that is friend of the project and pays the membership fee) * how a member loses the board or full member rights * how the board is supervised o when and how the reports of the board are presented o when and how the general assembly meeting takes place o how the reports are approved by full members * how to change the statute over time ... and perhaps more, but if we start and put some effort we can have it ready in reasonable time, avoiding trust and reliability issues in long term. Let's have debates here and contribute the results to the wiki. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] project status and my decisions
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello all, following the last events everybody perhaps wonders what is going on. I was surprised by the way everything happens and I am not sure yet how this resulted. Probably a bit of time is needed to clarify relations between developers themselves and community as well. There is not any official statement that a developer quited the support for this project, but I guess the trust between some is at lowest level possible. It is clear that we cannot say anything about a board right now that manages the project. From board discussions summary you may see that discussions were conducted to create a legal entity around the project, a non profit foundation. This might be hard to achieve in this status, but as a personal decision, I am not going in any other management group unless there is a legal framework backing up. Human relations prove to be pretty unpredictable and not conforming with initial statements. What is sure now - openser 1.3.x and 1.4.x will be maintained properly and I am volunteering to take care but hope others will join. It is really a good achievement of a sip server. Decision about future development will be based on community and developers feedback for this project. I am not going to promise anything by myself as I admit one person (to be more specific: myself) cannot take care of the entire code base -- I am just realistic, not willing to sell dreams. I think the people that still believe in the project can come with suggestions and alternatives of future development. What I can tell for sure, I am not going to join a fork of the original openser project. If something happened to make developers not working together, solution to solve nicely could have been found, in the worst, maybe just by splitting teams, but keeping old domain together just to show the new alternatives. "Why not joining a fork in this manner?"> Because I was one of the founders of this one and involved in development and management at top level. If I failed to build something I started, I had all the tools in my hands, why just creating something similar saying the old is crap but this new one will be the best? What is the difference? Why I didn't do it first time? Would I be able to fool someone? Does leaving my ship in secrecy on back door will bring more trust on me? I doubt and perhaps you, too. I am certainly assuming share of the failure in managing this project. I was there, in management, period. There is no single person responsible for mis-management, but it comes just easy for some to blame the others. But I am really confident we built outstanding sip server, all of us contributed with code, documentation, testing to an application that routes million and millions of minutes per day. There were accuses of bad things about project, I will argument in a separate message, this will get too long. So, I am asking: - how do you see a proper management structure for the future - feedback about future involvement, suggestions and alternative for going on Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] Register to SER with UDP
by Steven Wu 15 Aug '08

15 Aug '08

Hi All, Does SIP server at iptel.org support UDP register? When my UA rigister to iptel.org whith UDP, the 401 response always replied by SER at port 100xx(example 10028). But in my contact of rigister port assigned to 5060. So, my UA can't recive the 401 response, register failure. But use TCP everyting is OK? Can anyone give me help? Best regards, Steven Wu

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[Kamailio-Users] KAMAILIO v1.4.0 is released
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

KAMAILIO v1.4.0 is released August 7, 2008 Kamailio is the new name of the SIP server known so far as OpenSER. Due to trademark issues, the OpenSER project had to change the name. Officially, the new name was announced on July 28, 2008. Version 1.4.0 brings many new features since OpenSER v1.3.0 as well as improvements to existing components. An important share of effort was directed to code cleanup and stability. Moreover, new features in core and old modules were added while other 6 new modules were introduced. Download: You can download the tarball of the released sources at http://www.kamailio.net/pub/kamailio/latest/src/ Binary packages for several distributions you can find at http://www.kamailio.net/pub/kamailio/latest/bin/ http://www.kamailio.net/pub/kamailio/latest/packages/ Packages will be uploaded as soon as they are submitted. Documentation: * modules' documentation: http://www.kamailio.net/docs/modules/1.4.x/ * migration guide : http://www.kamailio.net/dokuwiki/doku.php/install:1.3.x-to-1.4.0 Summary of the new stuff in core: * overhauled DB API - better performance and safety, more common functionality integrated in the core * extensive cleanups in database drivers - integrate common functionality into the core for more stability and maintainability * SDP parser - provides an internal API for parsing SDP * fixup functions - provides an internal API for fixing module parameters * new free fixup functions - making safer to use some functions exported by modules in embedded languages such as Perl * extension to module interface - its now possible to use up to 6 parameters in module functions New modules: * db_oracle - ORACLE DB driver - connect to Oracle SQL Server * dialplan - Dialplan management - dialplan regular expression based translations * nat_traversal - NAT traversal helper * peering - RADIUS based peering allowing SIP providers to verify via a broker if source or destination request is from a trusted peer * ratelimit - SIP ttraffic shaping control and server load control based on your rules * userblacklist - User-specific blacklists New in existing modules: * acc module * it is possible to log accounting related output to a different log facility * cfgutils module * new FIFO functions get_config_hash and check_config_hash for config file checks * shared PV exported to configuration script * debugging helpers usable from config script for abort, shm_status and pkg_status * mysql module * enable timeouts on connect, read and write to prevent blocking on errors * database modules * renaming of all database modules, addition of the prefix 'db_', e.g. mysql is now db_mysql * carrierroute module * extension: much more flexibility in routing and database supported failure routing, improved internal structure * dialog module * internal API reworked for better flexibility * direction of the message provided via the internal API * new dialog callback types * new mi command: dlg_list_ctx * dispatcher module * ability to load destination URI flags from database at startup or reload * new algorithm to hash the content of a pseudo-variable * enum module * enum_fquery([...]) replaced by enum_pv_query([...]) * msilo module * notification system refurbished - message body, content-type, from address and contact header can be dynamically specified with pseudo-variables * pdt module * accepts now same destination domain bound to different prefixes * internal structures and logic optimized for memory and performance * lcr module * refactor module functions * pseudo-variable support * mediaproxy module * update to version 2.0 * better performance and scalability as packets are forward in kernel space New testing suite * a testing suite to ensure quality checking and regression tests reports has been developed, included in the repository and used during this development cycle Other important changes * modules' documentation has been migrated to DOCBOOK XML format * doxygen documentation extended and developer guide published * serweb tables have been removed * command line tool to manage dbtext in a SQL fashion These are not all, there is a big change log that gives more details: * http://www.kamailio.net/pub/kamailio/latest/ChangeLog More on http://www.kamailio.net Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] As a board member
by Adrian Georgescu 15 Aug '08

15 Aug '08

Hello, Because of sheer incompetence and abuse conduct of one of the recently adopted board members I decided to put my activities as a management board member of OpenSER/Kamailio project on hold. I will be able to resume my activities after the voluntary leave of Henning Westerholt. For those who do not know who I am, I was involved with OpenSER from Day 0 of its inception. Kind regards, Adrian

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[Serusers] SIP conference.
by Peter Cai 15 Aug '08

15 Aug '08

Hi all! I know H323 support conference but SIP says nothing about it. I want to make some study on this subject but there seems to be little information on Internet. Could some one give me some clues? Is there any RFC draft on this concept?

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[Kamailio-Users] Voicemail implementation...
by Gerard A. Matthew 14 Aug '08

14 Aug '08

Ok, I'm trying to get the voicemail system working, however for some reason my Asterisk box is receiving a duplicated IP Address on the c line from OpenSER....below is the sample sip header. Is there anything in my openSER config file that would cause such a duplication? Rgds, Gerard INVITE sip:u5073@<DOMAIN>:5061 SIP/2.0 Record-Route: <sip:<DOMAIN_IP>;ftag=1d498377;lr=on> Via: SIP/2.0/UDP <DOMAIN_IP>;branch=z9hG4bKaed4.96adae34.1 Via: SIP/2.0/UDP 192.168.1.112:51006;received=<CLIENT_XIP>;branch=z9hG4bK-d8754z-0127430e81514e54-1---d8754z-;rport=51006 Max-Forwards: 69 Contact: <sip:0078@<CLIENT_XIP>:51006;nat=yes;nat=yes> To: "5073"<sip:5073@<DOMAIN>> From: "G Laptop"<sip:0078@<DOMAIN>>;tag=1d498377 Call-ID: NDZiOTUyNWFmYTQ3M2JlMjNiYzdmZjcyMGYyZDljZWE. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 481 P-hint: usrloc applied v=0 o=- 6 2 IN IP4 192.168.1.112 s=CounterPath X-Lite 3.0 *c=IN IP4 <XXX.XX.XXX.XXX><XXX.XX.XXX.XXX>* t=0 0 m=audio 3649036490 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 1 : 4IoIRnGz iYQAZz1E 192.168.1.112 16720 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes

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[Kamailio-Users] Translate name in RURI to number
by Ali Jawad 14 Aug '08

14 Aug '08

Hi all..my users are all registered with names...I want to forward their calls to my pstn gw..however those only take numerical ANI numbers. So I need to change the sip packet on the Openser server to a number (diff number per user) before forwarding that to the PSTN Gateway ..any ideas ? Would ENUM work ? THanks

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[Kamailio-Users] [OT] Any domain registrar offering SRV, NAPTR and AAAA?
by Iñaki Baz Castillo 14 Aug '08

14 Aug '08

Hi, first of all I'm sorry for the OffTopic. I'm looking for a domain registrar and nameserver that offers me SRV, NAPT and AAAA registers ni their domain control panel. Unfortunatelly in Spain *NO ONE* offers it. Is there any one "outside"? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Kamailio-Users] Kamailio & CDRTool 6.5.XX & MediaProxy 2.0
by luzango mfupe 14 Aug '08

14 Aug '08

Hi guys,I read the release note of MediaProxy 2.0 that to be able to have it fully functioning one needs CDRTool version 6.5.X and above i have been searching all over with no avail can someone point where it if found?? by the way i just found the old version 6.4.1 in the author's site wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] new developer: Carsten Bock
by Daniel-Constantin Mierla 14 Aug '08

14 Aug '08

Hello, I wish to welcome long time contributor, Carsten Bock, among developers. He is going to be officially co-maintainer of the dispatcher module and help with several other modules such as dialog, uac, ... He has submitted lot of new features in the past to many parts of the code in openser. His id is 'carstenbock'. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] openser 1.2.1 errors in the log
by toly 14 Aug '08

14 Aug '08

Greetings, This is openser 1.2.1 compiled with TLS support. TLS is not turned on Did anybody experienced with the errors listed below: 08/13 16:28:05 13648 debug <tcp_main.c: 718> tcp_send: no open tcp connection found, opening new one 08/13 16:28:05 13651 err <tcp_main.c: 291> ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 08/13 16:28:05 13651 err <tcp_main.c: 459> ERROR: tcpconn_connect: tcp_blocking_connect failed 08/13 16:28:05 13651 err <tcp_main.c: 721> ERROR: tcp_send: connect failed 08/13 16:28:05 13651 err <../../forward.h: 105> msg_send: ERROR: tcp_send failed 08/13 16:28:05 13651 err <t_fwd.c: 767> ERROR:tm:t_forward_nonack: sending request failed 08/13 16:28:05 13651 debug <t_funcs.c: 255> ERROR:tm:t_relay_to: t_forward_nonack returned error Once I start getting these, proxy will become pretty much un-usable, I have to restart it. Thanks a lot. Hippo -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp18973275p18973275.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Error: driver error on query: Unknown column 'str_doc_col' in 'field list' ??
by David Loh 14 Aug '08

14 Aug '08

Hi guys, Since 1.4.0 released on Kamailio, I've been testing it on my R&D server. With the introduction of RLS, I'm trying to implement this module along with presence on my box, however, I'm having this error messages whenever rls_handle_subscribe() been called. ------------------------------------------------------------------------ Aug 14 00:06:03 localhost /sbin/kamailio[5139]: INFO: [sip:davidloh@192.168.72.128](192.168.72.1) - [eyeBeam release 1100l stamp 46319] entered Aug 14 00:06:03 localhost /sbin/kamailio[5139]: DBG: Request [SUBSCRIBE] from [sip:davidloh@192.168.72.128] (192.168.72.1) Aug 14 00:06:03 localhost /sbin/kamailio[5139]: _ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'str_doc_col' in 'field list'_ Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:core:db_do_query: error while submitting query Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:get_resource_list: while querying table xcap for [uri]=sip:davidloh@192.168.72.128 Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:rls_handle_subscribe: while attepmting to get a resource list Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:rls_handle_subscribe: occured in rls_handle_subscribe ------------------------------------------------------------------------ Have been google around for the solution but couldn't find any. Appreciate if anyone could enlighten me on this. Meanwhile, I was wondering if any of could enlighten me on how to get Kamailio + Presence + RLS works? No matter how I configure, SIP client (eyeBeam) seems not able to pull out any buddies-list/anything right from XCAP (integrated XCAP from Kamailio). Following is my configuration (unnecessary part have been striped) ------------------------------------------------------------------------ loadmodule "sl.so" loadmodule "tm.so" loadmodule "db_mysql.so" loadmodule "presence.so" loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 0) modparam("presence_xml", "integrated_xcap_server", 1) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "server_address", "sip:192.168.72.128:5060") loadmodule "imc.so" loadmodule "mi_xmlrpc.so" modparam("mi_xmlrpc", "log_file", "/var/log/kamailio-xmlrpc.log") modparam("mi_xmlrpc", "port", 8080) loadmodule "pua.so" loadmodule "rls.so" modparam("rls", "integrated_xcap_server", 1) modparam("rls", "xcap_root", "http://192.168.72.128/xcap-root:8080") modparam("rls", "rls_event", "presence") modparam("rls", "to_presence_code", 10) loadmodule "xcap_client.so" modparam("xcap_client", "xcap_table", "xcap") modparam("xcap_client", "query_period", 50) modparam("presence|presence_xml|imc|pua\rls", "db_url", "mysql://myuserid:mypassword@localhost/kamailio") ........... route[0] { .... if (is_method("NOTIFY")) { pua_update_contact(); route(4); } else if (is_method("PUBLISH|SUBSCRIBE")) { route(5); } route[5] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { $var(ret_code) = rls_handle_subscribe(); if($var(ret_code)==10) { handle_subscribe(); } t_release(); } else if(is_method("NOTIFY")) { rls_handle_notify(); xlog("L_INFO", "DBG: RLS handle_notify()"); } exit; } ------------------------------------------------------------------------ Did I missed anything ? Thanks, David Loh

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[Kamailio-Users] Non-numeric prefixes in carrierroute:scan_prefix
by mayamatakeshi 14 Aug '08

14 Aug '08

Hello, I just need a confirmation. We have Kamailio 1.3 running in production and it is working fine. But we received some reports from our OAM team that sometimes they want to set a non-numeric prefix at carrierroute::scan_prefix (like 'abc.') and this seems to not work. I have no complete details but it seems cr_user_rewrite_uri will fail if scan_prefix holds a non-numeric prefix. So I just like to know if scan_prefix is supposed to work with non-numeric prefixes or not. Regards, takeshi

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[Serusers] SER with Radius Authentication
by Samuel Muller 13 Aug '08

13 Aug '08

Hello all, I'm looking for an updated sample of "ser.cfg" who can help me about the use of Radius for authentication. Since release 2.0.0, it seems the function "www_challenge" does not exist ... If anyone has a config part concerning Register + Auth with Radius, it could be very helpfull ! In second point, I just wanna know if a database is always needed for multi-domains environment ? I've some problems with functions like lookup_domain. Many thanks, Samuel MULLER

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[OpenSER-Users] Problem getting redirect reason in uac_redirect
by Ruchir 13 Aug '08

13 Aug '08

I'm using uac_redirect module to handle redirect and accounting from openser. Redirect works fine so as cdr but I'm not getting how to write redirect reason in cdr. I didn't find any way to find and store redirect reason(call forward, busy, no answer) in CDR. Does anyone know how to do it?

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[Kamailio-Users] ERROR:uri_db:mod_init: Invalid table version of the subscriber table (Kamailio ver. 1.4)
by Rafael Risco Gonzales-Vigil 13 Aug '08

13 Aug '08

Hi I am trying to run new kamailio 1.4 but have this errors: *Aug 11 12:56:44 [20808] ERROR:uri_db:mod_init: Invalid table version of the subscriber table Aug 11 12:56:44 [20808] ERROR:core:init_mod: failed to initialize module uri_db Aug 11 12:56:44 [20808] ERROR:core:main: error while initializing modules* I have created new database 'kamailio' using kamdbctl and 'version' table has this values: table_name: subscriber table_version: 6 any idea? rafael PS: same problem with opensips 1.4...

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[Kamailio-Users] listen the new name
by Daniel-Constantin Mierla 13 Aug '08

13 Aug '08

Hello, we did some recordings of the new name, letting you know how some people pronounce the word. The audio files are linked from the first page http://www.kamailio.net/ or can be found directly at: http://www.kamailio.net/pub/kamailio_pronounciation/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] project foundation
by Daniel-Constantin Mierla 12 Aug '08

12 Aug '08

Hello, On 08/12/08 16:39, Adam Linford wrote: > Good stuff! > > I think that, for the most part, the business and technical facets of > the organisation should remain fairly distinct. The board needs > representation with regards to technical direction (and should have > this input from the organisations official 'members' under my > understanding of the current proposal), but its purpose should be to > keep developers free to code, and users free to use, safe in the > knowledge that resources will be made available (where neccessary) for > the project to achieve its objectives (both long and short term). yes, there are going to be no restrictions in usage or coding, but have an organization with legal framework behind that can take care of trademarks, manage resources and administration of the project. > > We could make non-executive positions on the board available. A good suggestion. > These should be unbiased people.. perhaps invite some big names in the > SIP field, that can provide an independent review of the boards > activities, as well as assisting the board in its role. This is a > tried and tested directorial structure that would also appeal to the > corporate world, which in turn might make it easier to achieve > corporate sponsorship/contributions/membership (as well as adoption of > the software in industry!). > > I don't think an executive committee is needed within the board.. the > board itself should suffice. If it doesn't, we don't have the right > board. I would suggest keeping the main board size small (5-8) to > keep it agile whilst maintaining good representation of members I think would be good to have some clear positions so people should know who to address to when comes to dealing with the foundation. Also, by this, people can allocate the proper time in their schedule for the tasks. We may avoid in this way some members expecting the others to do some work. > > Election rules need discussing.. duration of term for board and also > the achievement of 'member' status.. we dont want potential developers > discouraged from working with us in the traditional open source model > because they would have too much responsibility placed on them to be > 'active members'. Being an active member is just free will. But we want to avoid being a member with voting right just by paying some money. That meber should show interest and contribute to the project in a positive manner. Indeed, this needs to be addressed. > > Frequency of meetings etc are all relatively easy decisions that can > be deferred to the board once i elected. Yes, I believe the same. Cheers, Daniel > > Cheers, > Adam > > On 12 Aug 2008, at 12:16, Daniel-Constantin Mierla wrote: > >> Hello everybody, >> >> I think it is the time to move forward with project management >> structure. During last days there were talks and discussions on the >> lists, the solution to make the project independent and long term >> trustable is to create a foundation. >> >> We made a draft of purpose and organization structure and we want to get >> your comments. >> >> http://www.kamailio.net/dokuwiki/doku.php/admin:foundation-draft >> >> For the statute we are looking at similar cases of foundations around >> open source projects. Considering the developer population and >> contributors, I find as good and reliable place to register the >> foundation to be Germany. >> >> There is still long way to go and a lot to discuss: >> >> * >> >> membership fees >> >> * >> >> how one can become a full member (associate member can be anyone >> that is friend of the project and pays the membership fee) >> >> * >> >> how a member loses the board or full member rights >> >> * >> >> how the board is supervised >> >> o >> >> when and how the reports of the board are presented >> >> o >> >> when and how the general assembly meeting takes place >> >> o >> >> how the reports are approved by full members >> >> * >> >> how to change the statute over time >> >> ... and perhaps more, but if we start and put some effort we can have it >> ready in reasonable time, avoiding trust and reliability issues in long >> term. >> >> Let's have debates here and contribute the results to the wiki. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] branch_route not always called
by Alex Hermann 12 Aug '08

12 Aug '08

Hello again, The uac_replace_from issue led me to another problem. The branch_route is not always called even though I set it using t_on_branch everytime from REQUEST_ROUTE and FAILURE_ROUTE. The failing scenario is as follows: The call is routed to 2 domains (A,B) each with 2 SRV records (1,2). t_on_branch relay domainA branch domainA-1 - timeout branch domainA-2 - 480 failure route append_branch t_on_branch relay domainB branch domainB-1 - timeout branch domainB-2 - 200 OK ==> this branch doesn't call the branch_route Any idea's ? -- Greetings, Alex Hermann

2 3

[Serusers] problem error SER
by Bimantara Jayadiputro 12 Aug '08

12 Aug '08

hello sir i'm bimbim, student from indonesia now i'm trying to finish my final project my final project is about SER i have trouble with that when i start /etc/init.d/ser start there is an error with assigning addres below is the configuration of my ser ------------------------------------------------------------------------------------------------------------------------ cnc9@cnc9-desktop:/var/run/ser$ sudo /etc/init.d/ser start Starting ser: serListening on udp: 10.14.200.95 [10.14.200.95]:5060 udp: 2001:660:4701:1:0:0:0:1 [2001:660:4701:1:0:0:0:1]:5060 udp: sip4.sipv6.net [10.14.200.101]:5060 udp: sip6.sipv6.net [2001:660:4701:1:0:0:0:2]:5060 tcp: 10.14.200.95 [10.14.200.95]:5060 tcp: 2001:660:4701:1:0:0:0:1 [2001:660:4701:1:0:0:0:1]:5060 tcp: sip4.sipv6.net [10.14.200.101]:5060 tcp: sip6.sipv6.net [2001:660:4701:1:0:0:0:2]:5060 Aliases: tcp: localhost:5060 tcp: cnc9-desktop.local:5060 udp: localhost:5060 udp: cnc9-desktop.local:5060 stateless - initializing 0(0) . cnc9@cnc9-desktop:/var/run/ser$ Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 * 0(0) ERROR: udp_init: bind(6, 0x80f8d7c, 16) on 10.14.200.101: Cannot assign requested address* ------------------------------------------------------------------------------------------------------------------------ i also attached my ser.cfg i hope you can help me resolve this problem thank you

3 2

[Kamailio-Users] uac_replace_from and lumps again
by Alex Hermann 12 Aug '08

12 Aug '08

Hello, i'm still fighting with the very annoying lump model that OpenSER uses. I need to send out INVITE's with different From headers in a failover situation. More specifically I need to send out each step -original request and failure_routes- with a different From header. If any request forks (parallel or serial due to DNS SRV or registered contacts), it needs to keep current (same) the replacement From on all branches. I tried the following: try 1: REQUEST_ROUTE: uac_replace_from(a) t_relay(); FAILURE_ROUTE: append_branch() uac_replace_from(b) t_relay() the second uac_replace_from always produces garbage in the From try 2: REQUEST_ROUTE: t_newtran() uac_replace_from(a) t_relay(); FAILURE_ROUTE: append_branch() uac_replace_from(b) t_relay() Now the second uac_replace_from is correct, but when the first t_relay() does dns based failover, the original from is used for the second branch instead of the replacement. If the response code is locally generated 408, the second uac_replace_from produces garabage too. Is someone aware of a working solution for this (very common) situation? -- Greetings, Alex Hermann

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[Kamailio-Users] Need Help Configuring OpenSer with OpenXCAP
by Noble Antony T 12 Aug '08

12 Aug '08

Hi, I could configure Openser as RLS, and installed OpenXCAP. I wanted to use OpenXCAP with EyeBeam Client. (version 1.5.10.2 build 33793) OpenXCAP is giving the following error. Invalid Document Selector xcap-root(a)192.168.176.150/pres-rules/users/user1/presrules (the Document Selector context must be 'users' or 'global': 'pres-rules') Invalid Document Selector xcap-root(a)192.168.176.150/resource-lists/users/user1/resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') Invalid Document Selector xcap-root(a)192.168.176.150/pres-rules/users/user1/presrules (the Document Selector context must be 'users' or 'global': 'pres-rules') Invalid Document Selector xcap-root(a)192.168.176.150/resource-lists/users/user1/resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') UE is getting 404 Error. It seems the OpenXCAP is expecting 'users' or 'global' instead of 'resource-lists'. Can somebody please point out, if I am missing something..? I am attaching my Config Files along with this. Somebody please help me... -- Thanks & Regards, Noble Antony Thattil.

1 1

[Kamailio-Users] fr_inv_timer increased to 120 secs
by Kaeman Chris 12 Aug '08

12 Aug '08

Hello, What are the problems in production that can be encountered if the fr_inv_timer is increased from 30-45 secs to 120 secs for a sip-sip call? Anyone have found any issues? Thanks! KC --------------------------------- Unlimited freedom, unlimited storage. Get it now

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Re: [Kamailio-Users] ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2
by Daniel-Constantin Mierla 12 Aug '08

12 Aug '08

Hello, On 08/12/08 10:30, mahesh.peddi(a)info-spectrum.com wrote: > Hi Daniel, > > Thanks for your prompt reply.The output of "show create > table version" is as follows > > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > | Table | Create Table | > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > | version | CREATE TABLE `version` ( > `table_name` varchar(32) NOT NULL, > `table_version` int(10) unsigned NOT NULL default '0' > ) ENGINE=MyISAM DEFAULT CHARSET=latin1 | > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > 1 row in set (0.03 sec) > > Kindly guide me on this. the table structure is ok. Could you make sure the mysql module is compiled again if you updated libraries or so. I was pointed that there was a change in naming the INT type in mysql libraries -- maybe others on the list can jump quicker here with a yes or no. Cheers, Daniel > > Thank you, > > > Regards, > Mahesh Peddi > Infospectrum India Pvt. Ltd. > Cell: +91 9765775176 > IP-Phone Ext. - 764 > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: <mahesh.peddi(a)info-spectrum.com> > Cc: <users(a)lists.kamailio.org> > Sent: Monday, August 11, 2008 9:41 PM > Subject: Re: [Kamailio-Users] ERROR:presence:mod_init: 1Wrong version > v-1 for table <presentity>, need v2 > > >> Hello, >> >> please post the output of: >> >> show create table version >> >> in the mysql client. The errors shows some issue with the column type. >> >> Cheers, >> Daniel >> >> On 08/11/08 17:12, mahesh.peddi(a)info-spectrum.com wrote: >>> Hi All, >>> >>> I am working on Openser 1.3.2 version with no_tls support. >>> When i include >>> "modparam("presence", >>> "db_url","mysql://openser:openserrw@localhost/openser")" >>> >>> for the presence and run openserctl moni i am getting the error as >>> >>> ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo >>> ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", >>> "/tmp/openser_fifo")' in your config >>> ERROR: and also have loaded the mi_fifo module. >>> >>> And the errors in /var/log/messages >>> >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:table_version: >>> invalid type (2) or nul (0) version columns for presentity >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:presence:mod_init: 1Wrong >>> version v-1 for table <presentity>, need v2 >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:init_mod: failed to >>> initialize module presence >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:main: error while >>> initializing modules >>> Aug 10 09:37:41 rhel4 openser[5530]: NOTICE:presence:destroy: >>> destroy module ... >>> Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session opened for user >>> root by (uid=0) >>> Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session closed for user >>> root >>> >>> And when i check in the data base its showing the version as >>> >>> mysql> select * from version where table_name="presentity" ; >>> +------------+---------------+ >>> | table_name | table_version | >>> +------------+---------------+ >>> | presentity | 2 | >>> +------------+---------------+ >>> 1 row in set (0.00 sec) >>> >>> Please help me how to resolve this. >>> And as mention in the forums i modify the res.c as below shows even >>> i am facing the same error. >>> >>> >>> --------------------------------------------------------------------------------Index: >>> modules/mysql/res.c===================================================================--- >>> >>> modules/mysql/res.c (revision 4448)+++ modules/mysql/res.c (working >>> copy)@@ -103,7 +103,10 @@ break; default:- RES_TYPES(_r)[i] = >>> DB_STRING;+ LM_WARN("unhandled data type column (%.*s) type id (%d), >>> "+ "use DB_STRING as default\n", RES_NAMES(_r)[col]->len,+ >>> RES_NAMES(_r)[col]->s, fields[col].type);+ RES_TYPES(_r)[col] = >>> DB_STRING; break; } >>> }--------------------------------------------------------------------------------I >>> am enclosing my "openser.cfg" and "openserctlrc"Regards,Mahesh >>> PeddiInfospectrum India Pvt. Ltd.Cell: +91 9765775176IP-Phone Ext. >>> - 764 >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Re: [Kamailio-Users] poll: project name
by Helmut Kuper 12 Aug '08

12 Aug '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Klaus, Of course I agree. It's a matter of time to know what Kamailio stand for. I think Apache is a nicer name than Kamailio, but anyway: I agree the name is mainly a question of personal taste and currently I don't like that taste - nevertheless, I think the name should be changed to a more handy one, just to round up the whole product. And sure: The name won't stop telcos from using the Software rather that project structure, roadmap and quality. regards and peace Helmut Klaus Darilion schrieb: | Helmut Kuper wrote: |> -----BEGIN PGP SIGNED MESSAGE----- |> Hash: SHA1 |> |> Hm, |> |> well, maybe I have to make it a bit more clear what I meant. Kamailio is |> pronounceable in german, but it is not easy in terms of smoothyness - |> like openser was or freeswitch is ... |> |> KAMAILIO hmmm no, not with the best will in the world: I have problems |> with it to say it without explaining what that name points to. I think |> it is not a good name to inspire confidence. Maybe it needs a slogan to |> explain a bit more :D | | IMO this is all a matter of time. Would you think that Apache is a web | server and Eclipse is an IDE and Asterisk is an IP PBX? Me not. But now, | if you tell one of these names to an IT guy they know what you are | talking about. Thus, in my opinion, we just need some trust in the | product - I do not believe the argument that telcos will use open source | only if the name indicates open/free/SIP.... Nobody uses a product just | because of its name. | | But I have to accept that others don't like name ...:-( | | regards | klaus | | -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkihQDkACgkQ4tZeNddg3dwrJgCdENGNK/PAid+ftRtvgUFyP9hY ZXAAoJ5ErmBLbKuxTRvp6WmWzudI1BS9 =cyPy -----END PGP SIGNATURE-----

1 0

[Kamailio-Users] Identifying branch in ONREPLY-ROUTE?
by Alex Balashov 12 Aug '08

12 Aug '08

What is the simplest way to identify the branch (by branch index) to which a reply belongs in an ONREPLY-ROUTE? I have parallel forking with my registrar and do passive call statekeeping. The problem is that it must be branch-aware; I need to know if, for example, I got a successful provisional reply on one branch of the call and a failure on the other, and deal with those separately. Right now my statekeeping information is keyed by Call-ID, but since that persists across two branches, obviously some sort of branch-identifying information has to be present. I've tried setting a persistent variable to the value of $T_branch_idx inside a BRANCH-ROUTE, but in ONREPLY-ROUTE that just shows up with the most recent branch index, regardless of which branch the reply actually came in on. I'm probably missing something quite obvious here, so I ask - how do I incorporate some form of branch awareness into ONREPLY-ROUTEs? I really do not care what form that is - be it branch index, or some other built-in value, as long as it is consistent. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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Re: [Kamailio-Users] simple question: log 180, 200 responses
by Alex Balashov 12 Aug '08

12 Aug '08

Ali Soltani wrote: > Hi Alex, > > Thank you for the response. > > I tried to use onreply_route block in the main route as well as in > route1 as follow but still can not log response messages (180, 200, ..) > > route > { > t_on_reply("1"); > ... > } > > route[1] > { > if (is_method("INVITE")) > { > t_on_reply("1"); > } > ... > } > > onreply_route[1] > { > xlog("(onreplyroute1) usr=$ru method=$rm request_port=$rp > received_port=$Rp original port=$op destination_port=$dp src_port=$sp"); > } You can only have one reply route per request. If you specify multiple ones, only the last one will take effect. The problem in this case is that route[1] will never be called. Here's what you want: route { if(is_method("INVITE")) t_on_reply("1"); } onreply_route[1] { ... } -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 1

[Kamailio-Users] free_fixup_function
by Julien mangeard 12 Aug '08

12 Aug '08

In the 1.4 release notes, I have read that: => new free fixup functions - making safer to use some functions exported by modules in embedded languages such as Perl I don't find any reference to "free_fixup_function" in the Perl module source code. Are you sure that this feature works for Perl module? Someone work on it? Best Regards, -- Julien Mangeard

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[Serusers] Fw: 20080812 - Need help with configuring SER as a call forwarder server
by Morteza Alizadeh 12 Aug '08

12 Aug '08

Dear Sir/Madam, My previous email has been held by your system due to its large volume. I removed the log file from it and resend it again. If you think, that logs of the SER would help you, please provide me the way to send it as well. Your help is much appreciated. --- Sincerely yours, Morteza Alizadeh Design Engineer Cell Phone: +60 (12)-319-7439 Email: malizadeh(a)emerj.com Morteza.Alizadeh(a)gmail.com Skype: mortezaalizadeh Block D-17-6, Gurney Height Condominium, Jalan Bukit Keramat, 54000 Kuala Lumpur, Malaysia ----- Forwarded Message ---- From: Morteza Alizadeh <malizadeh(a)e-merj.com> To: serusers(a)lists.iptel.org Cc: vazirif(a)e-merj.com Sent: Tuesday, August 12, 2008 10:09:35 AM Subject: 20080812 - Need help with configuring SER as a call forwarder server Dear Sir/Madam, I am trying to use and configure your SER in a manner to let us forward our landline calls to the public sip.broadvoice.com gateway. Unfortunately, I am completely mixed up with the way it should be setup. There are some questions that I would be really appreciated if you can answer them or forward it to a right person. Please consider that I have already fully configured your SER to handle internal IP call between SIP clients and also I tested the SER with the SIP clients which are behind NAT demonstrate its NAT module works fine. Here is my current setup parameter to access landline call: * SER IP address: 207.34.42.53 * SER domain: ubuntu-serv.emerj.net * SIP gateway: sip.broadvoice.com * Internal user ID: 10002 * BroadVoice contact: 7184122016 * Number wants to access through SER: 9178462343 What I have in my MySQL server is as follow: * Domain ubuntu-serv.emerj.net in table 'domain'. * User 10002 registered with domain ubuntu-serv.emerj.net in table 'subscriber'. The password for this account is the normal password I set for it in database. * User 7184122016 registered with domain sip.broadvoice.com in table 'subscriber'. The password for this account is the password has been given by BroadVoice to us. * Defined a record in table 'usr_preferences' contains the following information: +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ | | 9178462343 | | callfwd | sip:7184122016@sip.broadvoice.com | 0 | 2008-08-11 10:00:29 | +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ This is the information I am using for my call forwarding purpose. Whenever the user tries to dial 9178462343, this record blindly forward it to the BroadVoice.com. Here are my questions: 1. I monitored and captured the packet between SIP, SER and BroadVoice. For the normal call directly from SIP to BroadVoice and also between SIP client and SER, a kind of SIP REGISTER message is sent at initial time and it seems that all the rest messages is based on the session value that is initially exchanged. When I try to use my SER as a call forwarding server, and when my SIP client tries to dial, SER doesn't send REGISTER message for account to 7184122016 to the BroadVoice. I want to know whether it is possible for SER to do this REGISTER before sending INVITE message for 9178462343 or not. 1. Is the call forwarding is the correct solution for this issue or not? Maybe there are some other things that I am not aware of them due to I am really new to SER environment. 1. I also saw that SER doesn't overwrite the values of Contact, To, and From in the INVITE message properly and everything is sent to the BroadVoice with the values of the ubuntu-serv.emerj.net as a domain and also initial SIP contact that all of them are unknown for BroadVoice gateway. How should I tell SER to overwrite this values and keeps the track of the changes? I also attached my both ser.cfg and the logs of the SER with my email. I would be really appreciated if you assist me resolving this issue and helping me with any document or sample ser.cfg file showing me how to realize the idea. I am looking forward to you response. --- Sincerely yours, Morteza Alizadeh Design Engineer Cell Phone: +60 (12)-319-7439 Email: malizadeh(a)emerj.com Morteza.Alizadeh(a)gmail.com Skype: mortezaalizadeh Block D-17-6, Gurney Height Condominium, Jalan Bukit Keramat, 54000 Kuala Lumpur, Malaysia

1 0

[Kamailio-Users] A function to add uri/header parameters
by Iñaki Baz Castillo 11 Aug '08

11 Aug '08

Hi, for now, the "normal" way to detect if an in-dialog request/response beyonds to a natted dialog is by adding a uri parameter ("nat=yes") in the "Contact" header of the first request and response and macthing that parameter in the RURI of subsequent in-dialog requests. Typically this is done in this way: search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); Well, this mechanims (that is part of official OpenSer courses) is very RCF3261 non compliant: 1) In SIP a header field name is case insensitive but the above function wouldn't match a "CONTACT" of "contact" header names (even if they are 100% valid). 2) "Contact" header can appear abbreviated as "m" (or "M") but the above doesn't match it. 3) 1.1.1. Known Limitations search ignores folded lines. For example, search("(From|f):.*@foo.bar") doesn't match the following From header field: From: medabeda <sip:medameda@foo.bar>;tag=1234 A solution for points 1 and 2 could be using "subst" with insensitive flag enabled, but it could be very anti-friendly. So I wonder: wouldn't be useful a good funtion to add/read/remove header and uri parameters (in any header or uri) being RFC 3261 compliant (this is: case insensitive and allowing header name abbreviations)? -- Iñaki Baz Castillo

2 4

[Kamailio-Users] ISN routing
by Krunal Patel 11 Aug '08

11 Aug '08

Hi, I would like to implement isn routing in openser. I am new to isn routing. Please help me if anybody of you have idea about isn routing. Thanking in advance - Krunal Patel

3 2

[Kamailio-Users] ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2
by mahesh.peddi＠info-spectrum.com 11 Aug '08

11 Aug '08

Hi All, I am working on Openser 1.3.2 version with no_tls support. When i include "modparam("presence", "db_url","mysql://openser:openserrw@localhost/openser")" for the presence and run openserctl moni i am getting the error as ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. And the errors in /var/log/messages Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:table_version: invalid type (2) or nul (0) version columns for presentity Aug 10 09:37:41 rhel4 openser[5530]: ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2 Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:init_mod: failed to initialize module presence Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:main: error while initializing modules Aug 10 09:37:41 rhel4 openser[5530]: NOTICE:presence:destroy: destroy module ... Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session opened for user root by (uid=0) Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session closed for user root And when i check in the data base its showing the version as mysql> select * from version where table_name="presentity" ; +------------+---------------+ | table_name | table_version | +------------+---------------+ | presentity | 2 | +------------+---------------+ 1 row in set (0.00 sec) Please help me how to resolve this. And as mention in the forums i modify the res.c as below shows even i am facing the same error. --------------------------------------------------------------------------------Index: modules/mysql/res.c===================================================================--- modules/mysql/res.c (revision 4448)+++ modules/mysql/res.c (working copy)@@ -103,7 +103,10 @@ break; default:- RES_TYPES(_r)[i] = DB_STRING;+ LM_WARN("unhandled data type column (%.*s) type id (%d), "+ "use DB_STRING as default\n", RES_NAMES(_r)[col]->len,+ RES_NAMES(_r)[col]->s, fields[col].type);+ RES_TYPES(_r)[col] = DB_STRING; break; } }--------------------------------------------------------------------------------I am enclosing my "openser.cfg" and "openserctlrc"Regards,Mahesh PeddiInfospectrum India Pvt. Ltd.Cell: +91 9765775176IP-Phone Ext. - 764

2 1

[Kamailio-Users] naming of open source projects
by Klaus Darilion 11 Aug '08

11 Aug '08

Hi! I really do not understand why so many people think that the name should include the word "open" or should somehow include parts of the protocol (SIP) or its usage (server, proxy ...). If I take a look at very successful open source projects they do not follow this, e.g. Apache, Eclipse, Pidgin, Wireshark, Asterisk, Tomcat, Boost ... regards Klaus

3 2

Re: [Kamailio-Users] poll: project name
by luzango mfupe 11 Aug '08

11 Aug '08

"Free to Talk" Not bad at all i bet most of us here would prefer an English name!!! Why Kamailio is bad??? relax guys u will be able to pronounce it well soon... My choice iiis numberrr 2. On Mon, Aug 11, 2008 at 3:31 PM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: [Kamailio-Devel] poll: project name (Klaus Darilion) > 2. Re: poll: project name (Yazid Hadj Said) > 3. Re: poll: project name (Ardjan Zwartjes) > 4. Re: [Kamailio-Devel] poll: project name (David Villasmil) > 5. Re: [Kamailio-Devel] poll: project name ( I?aki Baz Castillo ) > 6. Re: poll: project name (Carsten Bock) > 7. Re: poll: project name (Alex Balashov) > 8. Re: poll: project name (Johansson Olle E) > 9. Re: [Kamailio-Devel] poll: project name (David Villasmil) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 11 Aug 2008 13:39:29 +0200 > From: Klaus Darilion <klaus.mailinglists(a)pernau.at> > Subject: Re: [Kamailio-Users] [Kamailio-Devel] poll: project name > To: miconda(a)gmail.com > Cc: users <users(a)lists.kamailio.net>, devel <devel(a)lists.kamailio.net> > Message-ID: <48A024F1.5030105(a)pernau.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > speaking as user: > 1) Yes, I love it. Please keep it. > > Daniel-Constantin Mierla schrieb: > > Hello everybody, > > > > some people expressed inconveniences regarding the new name, so we would > > like to get your opinion about, therefore this is a poll that will get > > us to a decision. Here is the question: > > > > Do you like the name "Kamailio"? > > 1) Yes, I love it. Please keep it. > > 2) It is OK. > > 3) I do not mind the name at all > > 4) I do not like it > > 5) I hate the name and we should change it > > > > There is no need of explanations, as it is up to personal taste and > > decision. Just vote by option. > > > > Cheers, > > Daniel > > > > PS. for those interested... > > > > As short history about renaming story. On the 28th of March the board > > was informed about issues with SER part of project name. You can easily > > find who has the trademarks. On the 7th of April the board was informed > > about an agreement for the deadline as of September 30, 2008 to use old > > name. From there on, all old releases, documents, etc. can be referred > > by the old name, but nothing new. Other few discussions about what > > should/could be kept/changed followed and on 12th of May started the > > discussion about the new name with two main direction, something > > similar, to remind about old project or just something completely new, > > to be used as brand name. Couple of proposals and discussions regarding > > possible names followed up to 16th of July when was the deadline and the > > start of vote. Majority was reached on 17th but it was required to have > > opinions from all members of the board, last came from Bogdan, on the > > 28th of July. > > > > All discussions about migration were on the board mailing list. No > > change were done to openser.org configuration, just the new domain with > > kamailio in it was pointed to same server. Every step done was reported > > to board, following a schedule discussed there. Another discussion was > > about making the renaming and 1.4.0 release in same day. Being > > practically impossible to do all the work in one day, it was decided to > > do first the announcement about renaming. That happened on 28th when all > > agreed with the new name. > > > > As you can see, there was no 5 months of renaming duration. There was no > > change in openser.org DNS configuration undertaken with/without the > > knowledge of all board members. The new name domains were reserved when > > majority was achieved, in any case; dns entries for new name were > > pointed to old server shortly after booking the domains. > > > > > > ------------------------------ > > Message: 2 > Date: Mon, 11 Aug 2008 13:52:02 +0200 > From: "Yazid Hadj Said" <yazid.hadjsaid(a)gmail.com> > Subject: Re: [Kamailio-Users] poll: project name > To: users(a)lists.kamailio.net > Message-ID: > <8dfa8c1a0808110452w274cb206te3160cbe4dd8fd2(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all, > > > >Do you like the name "Kamailio"? > >1) Yes, I love it. Please keep it. > >2) It is OK. > >3) I do not mind the name at all > >4) I do not like it > >5) I hate the name and we should change it > > I do not like it. > I think that at least the word "Open" must be included in the new name > > > Regards > Yazid >

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Re: [Kamailio-Users] simple question: log 180, 200 responses
by Yazid Hadj Said 11 Aug '08

11 Aug '08

Hi Ali, I think that you must call : if(status=="180") { xlog("this is a 180 Ringing reply"); }; if(status=="200") { xlog("this is a 200 OK reply"); }; in the onreply_route, note in tne route function

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[Kamailio-Users] simple question: log 180, 200 responses
by Ali Soltani 11 Aug '08

11 Aug '08

Hi, I am using openSER as a simple sip proxy for calls between two sip end-points. By snooping the traffic I can verify that the openSER proxies ALL sip messages/responses but non of the following log statements do not log the response messages (180, 200). Am I missing anything. route { xlog("(route) usr=$ru method=$rm request_port=$rp received_port=$Rp original port=$op destination_port=$dp src_port=$sp"); xlog("(route) reply_status=$rs"); if(status=="180") { xlog("this is a 180 Ringing reply"); }; if(status=="200") { xlog("this is a 200 OK reply"); }; ..... } Thanks,

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Re: [Kamailio-Users] poll: project name
by Yazid Hadj Said 11 Aug '08

11 Aug '08

Hi all, >Do you like the name "Kamailio"? >1) Yes, I love it. Please keep it. >2) It is OK. >3) I do not mind the name at all >4) I do not like it >5) I hate the name and we should change it I do not like it. I think that at least the word "Open" must be included in the new name Regards Yazid

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[Serusers] 404 Not Found
by dwianto dana 11 Aug '08

11 Aug '08

Hi, I tried to use SER openimscore and connect it with sailfin. Tried to register using X-Lite. What I get in the Core is SIP: Status:404 Not Found - destination user not found on this S-CSCF I am using a default user (Alice and Bob),, so it is kinda strange to get this information, user not found. Thanks for the answers given

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[Kamailio-Users] Req for Coments about my config
by Ali Jawad 11 Aug '08

11 Aug '08

Dear All I am working with some developers to create a sip client that supports TLS for me. During the beta stage they created the phone and it sent "sips" packets with transport=tls in the packet. The thing is that I used the sethostport and t_relay to send data to my PSTN gateway that works on UDP but in both cases I was not able to make PSTN calls. The developers created a custom openser.cfg that rewrites sips packets to sip and removes the transport=tls from the packets. How efficient is this ? Is this the typical approach to make a TLS call through Openser where the PSTN gateway does not support TLS ? Please have a look at the config file if you can. http://pastebin.com/m20f7d25a Thanks

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Re: [Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008
by Cesar Troncoso 11 Aug '08

11 Aug '08

Hello! I am also up for a meeting in Bucharest on Monday next week. It will be a good chance to get different points of view about the kamailio/opensips discussion and the way developers and board members think to move forward in the future. Regards, Cesar ------------------------------ Message: 6 Date: Tue, 05 Aug 2008 18:57:50 +0300 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: [Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008 To: users <users(a)lists.kamailio.org>, devel <devel(a)lists.kamailio.org>, business(a)lists.kamailio.org Message-ID: <4898787E.7020602(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello Everybody, as many expressed their intention for an openser/kamailio event in Bucharest, next week, I am going to organize a dinner in a place with nice Romanian beer, wine and food. Seeing the suggestions for the date, I am proposing August 11, 2008 (Monday) -- let me know if you have opinions. Exact hour and place will be sent as soon as I get an idea about number of participants. So far, expressed the intention to participate: - Henning Westerholt, developer and board member - Elena-Ramona Modroiu, developer and board member - Daniel-Constantin Mierla, developer and board member - Michiel ten Hagen - Sebastian Schuman Please write me if you want to attend, I need to know the number of attendants to make the reservation. The event is in the same fashion as the ones happening in Barcelona, Vienna, Berlin, ..., in the past -- everybody pays for its expenses. If you want to have some preamble (might not be same place, but...), you can check some photos from last similar event in Bucharest after the ROSDEV2008 conference: http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/index1.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] fr_inv_timer issue
by Kaeman Chris 11 Aug '08

11 Aug '08

Hello, We have an fr_inv_timer set to 45 secs. When i make calls to an 800 number, the call hits to the 1st provider on LCR and IVR recording starts asking to hit an option 1,2,3,etc & if the caller dont hit any option and leave the call up to 45 secs. Than fr_inv_timer kills the transaction by sending a CANCEL message and thereby openser forces the call to send it over to the fail over route. Is this correct should it try the call on the failover route because the 1st provider was able to complete the call than why the call went over to the failover route? here as soon as the call switches to the 2nd provider the IVR message starts again from the option 1 leaving the option 3 in the middle of the recording. is this correct? let me know if the query is not clear. Thanks in advance --------------------------------- Unlimited freedom, unlimited storage. Get it now

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[Kamailio-Users] Radius + openser
by Amit Jain 11 Aug '08

11 Aug '08

Hello everyone, I was looking for documentation for Radius AAA + Openser. I tried to google but the links provided are not working. i think since the site has been renamed. Can any provide me document if saved or any other related link. Link to which i was referring . http://openser.org/docs/openser-radius-1.0.x.html Regards, Amit Jain

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[Kamailio-Users] Radius + openser
by Amit Jain 11 Aug '08

11 Aug '08

Hello everyone, I was looking for documentation for Radius AAA + Openser. I tried to google but the links provided are not working. i think since the site has been renamed. Can any provide me document if saved or any other related link. Link to which i was referring . http://openser.org/docs/openser-radius-1.0.x.html Immediate help required :( Regards, Amit Jain

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[Kamailio-Users] Should really 't_on_branch' remain in serial forking?
by Iñaki Baz Castillo 11 Aug '08

11 Aug '08

Hi, imagine this simple case: -------------------------- route { t_on_branch("1"); t_on_failure("2"); xlog("We are in 'route'"); t_relay("1.1.1.1"); } branch_route[1] { xlog("We are in 'branch_route[1]'"); # do something.... } failure_route[2] { xlog("We are in 'failure_route[2]'"); append_branch(); t_relay("2.2.2.2"); } --------------------------- In case an error occurs when forwarding the request to 1.1.1.1 this would be the screen log output: We are in 'route' We are in 'branch_route[1]' We are in 'failure_route[2]' We are in 'branch_route[1]' This is: branch_route[1] will also be runned **again** after the failure route since it was loaded in the first forward attemp. I really don't know if this is intuitive or not. The only way to "dissable" branch_route[1] in the failure route is by adding: ------------------- failure_route[2] { t_on_branch("2"); # <--- Dissable t_on_branch("1") xlog("We are in 'failure_route[2]'"); append_branch(); t_relay("2.2.2.2"); } branch_route[2] { xlog("We are in 'branch_route[2]'"); # Nothing to do here. # This route is neccesary to dissable t_on_branch("1") } ------------------ This would show: We are in 'route' We are in 'branch_route[1]' We are in 'failure_route[2]' We are in 'branch_route[2]' What about if "t_on_branch" wouldn't remain loaded after a failure route or serial forking? wouldn't be more intuitive to re-enable it explicitely when required? -- Iñaki Baz Castillo

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[Kamailio-Users] Res: serMyAdmin login problem invalid username and/or password
by Flavio Goncalves 10 Aug '08

10 Aug '08

Hi Muhammad, I will check the problem this week and return to you soon. Regards, Flavio ----- Mensagem original ---- De: muhammad akl <muhammad.akl(a)gmail.com> Para: Flavio Goncalves <flavio(a)asteriskguide.com> Cc: users(a)lists.kamailio.org Enviadas: Domingo, 10 de Agosto de 2008 12:44:47 Assunto: Re: [Kamailio-Users] serMyAdmin login problem invalid username and/or password thx Flavio for ur response but i still have another question the main problem with the user "admin@localhost " and password "openserrw" that were created by the openser_mysql.sh is still happening got the same message "invalid username and/or password " but when i removed that account from the subscriber table and used the openser.sql file from sermyadmin i have successfully logged in but after modifying the openser.sql file to match my environment so after that i tried to register with a new account from the home page using register button besides to login button and filled the account's information there everything worked fine but the problem happend here now is when i logged with the "admin@setup" and password "secret " account to approve the new account for users i got this message pastebin.com/m7154020b p.s for alot of testing i have used more than one version of serMyAdmin beginning with ver 0.3 to 0.7 !! sorry for my poor english as am not native BR muhammad

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[Kamailio-Users] Res: serMyAdmin login problem invalid username and/or password
by Flavio Goncalves 10 Aug '08

10 Aug '08

Hi Muhammad, Check the updated procedure at www.sermyadmin.org. The default login is asmin@setup, password secret as stated in the procedure. Flavio E. Goncalves ----- Mensagem original ---- De: muhammad akl <muhammad.akl(a)gmail.com> Para: users(a)lists.kamailio.org Enviadas: Domingo, 10 de Agosto de 2008 6:28:53 Assunto: [Kamailio-Users] serMyAdmin login problem invalid username and/or password hey all i'm trying to configure sermyadmin web interface to administrate openser-1.2.2 through it firstly i installed apache-tomcat and configure it successfully then i downloaded the sermyadmin.war and copied it to /usr/local/tomcat/webapps then downloaded mysql-jdbc and copied it to /usr/local/tomcat/lib also configure the context.xml and modified the Context element and added the user have full privileges to access openser databases then when tried to login with user "admin@locahost" and pass " openserrw" default values created by the openser_mysql.sh i got invalid username or password ? any suggestions ? serMyAdmin version used .7-BETA i also tried the howto found in sermyadmin.org but with no success by trying to use the prepared openser.sql file if anyone could i'd appreciate that

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[Kamailio-Users] How to revert changes in SDP during serial forking?
by Iñaki Baz Castillo 10 Aug '08

10 Aug '08

Hi, I have a scenario in which the call is first forwarded to user location using MediaProxy (so I modify the SDP in OpenSer) and if it gets a error response a failure route forwards the request to a media server with public IP for which I don't need and don't want to use MediaProxy. But since I change the SDP in the first forwarding (to the user) then I can't use the original SDP in the sequential forwarding to the media server. Is possible to do it? Thanks a lot. -- Iñaki Baz Castillo

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[Kamailio-Users] serMyAdmin login problem invalid username and/or password
by muhammad akl 10 Aug '08

10 Aug '08

hey all i'm trying to configure sermyadmin web interface to administrate openser-1.2.2 through it firstly i installed apache-tomcat and configure it successfully then i downloaded the sermyadmin.war and copied it to /usr/local/tomcat/webapps then downloaded mysql-jdbc and copied it to /usr/local/tomcat/lib also configure the context.xml and modified the Context element and added the user have full privileges to access openser databases then when tried to login with user "admin@locahost" and pass " openserrw" default values created by the openser_mysql.sh i got invalid username or password ? any suggestions ? serMyAdmin version used .7-BETA i also tried the howto found in sermyadmin.org but with no success by trying to use the prepared openser.sql file if anyone could i'd appreciate that

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[Kamailio-Users] OpenSIPS project
by Bogdan-Andrei Iancu 10 Aug '08

10 Aug '08

Hello everybody, A new project - OpenSIPs (Open SIP Server) - became active today, with its first release (OpenSIPS 1.4.0). What is OpenSIPS about? OpenSER History: ---------------- Voice System started the OpenSER project with a vision of running an open project that will deliver a professional software. Along the years Voice System was the main contributor and sustainer of the OpenSER project, organizing conferences, events and tutorials, all aiming to consolidate OpenSER as a top solution for industry VoIP. OpenSER Issues: --------------- In the past time, several critical problems related to OpenSER project were identified by us: - the need for a new reliable release - delayed or low-quality releases are affecting the credibility of the project - degradation of the project quality - testing, performance measurements and code quality control are overlooked - lack of the control, management and coordination of the project - critical project issues could not be handled or solved OpenSIPS Solutions: -------------------- OpenSIPS is a continuation of the OpenSER project - we have a moral obligation to develop and deliver the high quality and reliable software we envisioned when starting OpenSER. 1) a 6 months release cycle, but with no compromise for the technical issues 2) extensive testing and performance measurements before each major release 3) contributions are guaranteed to be answered and integrated (if accepted) in less than 1 month 4) code review, architecture control and integrated design for software quality 5) technical management with a strict developer hierarchy and developer cooperation to prevent deadlocks or un-availabilities 6) intellectual honesty by having decisions taken on value-based criteria and arguments. To find more about the project, see the official web page : http://www.opensips.org OpenSIPS 1.4.0 release: ----------------------- First major release of OpenSIPS is n**ow available. It incorporates new functionalities like * dialog profiling extension * local route for internally generated requests * dial plan / translation module * general NAT traversal module (for non-invite signalling) * peering and rate limit (traffic shaping) modules , enhancements like: * SRV load balancing (based on weights) * nonce re-usage verification (safer authentication) and bug fixing - code, specs and functionality bugs. The full Changelog is available on the web site: To get the OpenSIPS 1.4.0 version, see the download page at: http://www.opensips.org/index.php?n=Resources.Downloads Best regards, Bogdan

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[Kamailio-Users] 603 Declined
by Kaeman Chris 10 Aug '08

10 Aug '08

Hello, Does openser as soon as receives 603 Declined from the termination kills the call and not route it to the failover route? Is this correct? Thanks, KC --------------------------------- Unlimited freedom, unlimited storage. Get it now

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[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 09 Aug '08

09 Aug '08

I will be out of the office starting 08/08/2008 and will not return until 08/11/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

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[Kamailio-Users] Compiling MediaProxy 2.0 on CentOS 5
by Andrew Yager 09 Aug '08

09 Aug '08

Hi, Apologies if you have already seen this on the OpenSIPS users list; it seems that some of the domain changes are a little confusing for the moment. I'm trying to compile MediaProxy 2.0 on CentOS 5 for use with Kamailio. I've installed libnetfilter_conntrack from RPMForge, but I am getting an error with compiling. ./setup.py build running build running build_py running build_ext building 'mediaproxy.interfaces.system._conntrack' extension gcc -pthread -fno-strict-aliasing -DNDEBUG -O2 -g -pipe -Wall -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -m64 -mtune=generic -D_GNU_SOURCE -fPIC -fPIC - DMODULE_VERSION=2.0.3 -DIPTABLES_VERSION="1.4.0" -Imediaproxy/ interfaces/system/libiptc/include -I/usr/include/python2.4 -c mediaproxy/interfaces/system/_conntrack.c -o build/temp.linux- x86_64-2.4/mediaproxy/interfaces/system/_conntrack.o mediaproxy/interfaces/system/_conntrack.c: In function ‘ForwardingRule_init’: mediaproxy/interfaces/system/_conntrack.c:223: error: ‘NFCT_SOPT_SETUP_REPLY’ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:223: error: (Each undeclared identifier is reported only once mediaproxy/interfaces/system/_conntrack.c:223: error: for each function it appears in.) error: command 'gcc' failed with exit status 1 Current IPTABLES info: % iptables --version iptables v1.3.5 % lsmod | grep conntrack ip_conntrack_netbios_ns 36033 0 ip_conntrack 91237 1 ip_conntrack_netbios_ns nfnetlink 40457 1 ip_conntrack Is there a package I am missing, or am I simply fighting a loosing battle and need to use a different flavor of Linux to get it to work easily? Thanks, Andrew -- Andrew Yager, Managing Director (MACS BCompSc MCP) Real World Technology Solutions Pty Ltd ph: 1300 798 718 or (02) 9037 0500 fax: (02) 9037 0591 mob: 0405 152 568 http://www.rwts.com.au/ or http://www.stonebridgecomputing.com.au/

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[Kamailio-Users] failure route issue
by Mouncif Benniane 08 Aug '08

08 Aug '08

I am using openser 1.3, I can't make the failure_route to work properly. here is my config: modparam("tm", "fr_timer", 2) # set fr_timer route{ if (is_method("INVITE")) { avp_delete("*"); if (avp_db_query("SELECT prefix,grp_id,priority,custid,did,ip,port,failure_set,failure_i p,failure_port FROM test_table WHERE prefix='$rU'","$avp(s:prefix);$avp(s:grp_id);$avp(s:priority);$avp(s:custid);$avp(s:did);$avp(s:ip);$avp(s:port );$avp(s:failure_set);$avp(s:failure_ip);$avp(s:failure_ port)")) { $ru = "sip:"+$rU + "@" + $avp(s:ip) + ":" + $avp(s:port); if ($avp(s:failure_set) == y ) { $avp(i:3) = 3; t_on_failure("1"); t_relay(); } } } else { xlog("L_NOTICE", "avp_db_query() returned no records or failed query\n"); }; } ### END OF ROUTING failure_route[1] { # status is 408 if openser session timer fires if (t_check_status("408")) { $ru = "sip:"+$rU + "@" + $avp(s:failure_ip) + ":" + $avp(s:failure_port); t_relay(); } } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } So I set ip field to no responsive one and failure_ip to responsive one, but still the invite does not fail to the last one (failure_ip) after the timeout. Am I doing something wrong? cheers!!

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[Kamailio-Users] Kamailio in orange (logo submission).
by Alex Balashov 08 Aug '08

08 Aug '08

I have a logo submission from someone who works with me at Evariste - of artistic inclinations. I don't know what the proper submission path for the logos is, so I'm posting it here. Sorry if that's annoying. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] parameter 'ftag' and 'avp'
by Steven Wu 08 Aug '08

08 Aug '08

Hi all, What is meaning of ftag or avp parameter in Record-Route herder? SER inserted this Record-Route. As far as know, OpenSER is come from SER. Best regards, Steven Wu

2 1

[Kamailio-Users] comments to pointed openser issues
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello, these issues were pointed in the first place, my comments follow inline. 1) the need for a new reliable release - delayed or low-quality releases are affecting the credibility of the project - http://lists.kamailio.org/pipermail/users/2008-August/018999.html - the link above shows this one was a faster release than in past - http://www.kamailio.net/mos/view/KAMAILIO-v1.4.0-Released/ - the link above shows the efforts in creating a testing suite which was running on devel server during this development cycle - the link above shows no much new things in the core, but code cleanup, duplicity removal, source code documentation 2) degradation of the project quality - testing, performance measurements and code quality control are overlooked - presented before, testing suite was built and running, lot of code was reviewed and improved - who in the new project will do that and couldn't do it because was blocked in the old project 3) lack of the control, management and coordination of the project - critical project issues could not be handled or solved - I am not aware of mis-coordinations that could affect the project so badly - I am not aware of critical issues that could not be solved and there was no report of a critical deadlock - if there was a failure in management, all from the board share same fault Moreover, I believe there are facts in the past that openser took care of its quality. I repeat as I said that QA might not be enough no matter is done. If openser 1.3.x is the core component of a platform hosting about 2 millions of subscribers does not ensure some QA, then nothing else can be said, see: http://www.ilocus.com/2008/03/the_largest_voip_offering_base.html None above motivates such reactions as happened recently, but if does, then probably with some openess and discussions everything would have been better now. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] OpenSIPS vs. Kamailio
by Darren Sessions 08 Aug '08

08 Aug '08

I have to say, this is probably one of the harder decisions I've had to make in a very long time. I have to say the OpenSER / SER decision was much easier. While I obviously don't know all the details of what's gone on between Bogdan and Henning or whomever else, I just really have a hard time swallowing Hennings explanation of what happened and his, what seems to be, emotional reaction. More so than that, it seems like there were quite a few decisions made by the board without immediately letting the community know about it, even quite a while after these decisions were made. While I do not dispute Hennings technical ability, It is my opinion that his disrespectful attitude, his "shotgun" conduct, and his obvious lack of management ability is not befitting of someone on Kamailio project's board regardless of who is at fault here. Honestly, I'm a bit disappointed that Daniel hasn't said more as well. I met him at the fall VON in 2005 and he was indeed very pleasant and I enjoyed our conversation. All I've heard is things like 'not wanting to discuss any further', etc. - but the fact remains that wether it's a comfortable topic or not, there are quite a few of us that are using OpenSER in a commercial environment and these are things that we NEED to know in order to make an informed decision. In the end, all I know is now there are two projects and I have to choose which one. Based on what I've read in the change-logs, my direct experience with Bogden in a technical support role directly related to SER at a previous business - my company will be using OpenSIPS going forward. I have a very high regard for virtually all of the Kamailio / OpenSER group, and again, I am disappointed and absolutely hate that I've been forced into a position to have to make a decision like this. I will continue to be active on both sets of lists for the foreseeable future to monitor progress. Having said all that, I can't tell you how much I hope things can still be resolved and we can all have a unified project. I would implore the Kamailio board to do whatever it takes to bring Bogdan back into the group for the sake of the community. Hopefully something like that won't be just a dream. Thanks, - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com http://www.linkedin.com/in/dsessions _____________________________

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[Kamailio-Users] releases history
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello, as there were accuses about a delayed release, I am showing the dates when a major release was out: June 14, 2005 - first openser release, just a rebuild of ser 0.9.x October 28, 2005 - openser 1.0.0 July 10, 2006 - openser 1.1.0 (after more than 8 months) March 12, 2007 - openser 1.2.0 (after more than 8 months) December 13, 2007 - openser 1.3.0 (after more than 9 months) August 7, 2008 - openser 1.4.0 (less than 8 months) So, maybe my memory is damaged, but you should be able to learn it from mailing lists and web. I do not buy that motivation about project quality degradation in release management. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] openser board summary
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello guys, first I apologize for the situation we are in and for my silence in the last days as I tried to get the release out and not put gas on fire. The board looks like being now an evil thing, maybe it is our fault we haven't given more details about insights, but there were not many as ones could imagine and the decisions were presented to community. Here are the outlines of the discussions on the board, since September 2005, when first mail came there: 2005-09 DCM: decide release number => resulted in openser 1.0.0 DCM: business mailing lists => resulted in creation of the list 2005-10 AG: asks about making AG a trusted CA for default delivery of openser => self signed or something invalid to force users to change 2005-11 DCM: starts discussion about making an openser statute DCM: puts in dicussion the behavior of Harry on the mailing lists DCM: contacts Harry 2005-12 ALL: continuation of previous mail discussions 2006-03 DCM: plans for new release 2006-04 JH: discussion about presence implementation 2006-05 BIA: about project hosting: still sourceforge or move out ALL: Sourceforge problems discussions JH: presented some options for hosting 2006-06 AG: sip summer code JH: about 1.1.0 2006-07 JH: presence server AG: minutes if SIMPLE WG at IETF DCM: GPL issue with openssl and debian 2006-08 BIA: openser summit 2006-09,10 ALL: openser summit 2006-11 AG: history update in relation with old project BIA: TLS discussion 2006-12 ALL: history update 2007-01 KD: FOSDEM DCM: openser 1.1.1 BIA: openser at spring von 2007-02 BIA: openser donnations DCM: openser events 2007-04 BIA: Linuxtag 2007-09 DCM: summit 2007 2007-10 CS: retires from board BIA: proposes Henning => no decision as no rules for electing, makes that a point in todo BIA: openser 1.2.3 2007-11 DCM: openser business environment => hosting it within the website couldn't get a decision, posponed with alternative for dedicated site BIA: new developer Maxim Sobolyev 2008-03 BIA: on the 28th announces to board the issues with trademark, apparently some board member and other people were already aware AG: openser pages update 2008-04 ALL: agreed with rename - started the idea of a foundation to register trademark, takes on managing resources HW (not a member at that time but was included in the discussions related to TM) - proposes new developer Iouri 2008-05 BIA: on the 12th started the discussion about new name foreseeing 2 directions: something similar, to remember about the old project, or something totally different 2008-06 DCM: 9th: paypal report for last year => total donations to project's sourceforge account 52.05Euro (DCM Paypal account closed, the money still at DCM, proposed to use them for buying domains for new name) DCM: 9th: forwards a proposal received at end of April to team(a)openser.org from PacktPub to donate 2% or 5% (if we keep news on the site about the book) of openser book sales to the project. DCM asked for decision as there was no legal entity to deal with. (note: team(a)openser.org was created at the time openser forked in 2005, to get private feedback at that time, was used for the summit 2006 for contact, open for any post, includes the founders, about 10 good mails out of thousands spams per year) BIA: 9th: adds that in a discussion with PacktPub (not on team or board, as I haven;t seen such) proposed to give the 5% for openser devel course organized by him. The course was announced to the lists on the 4th of June. DCM: 9th: expressed surprise as he thought that was a different deal, made privately between BIA and PacktPub BIA: ask the board about donation, saying that actually the 5% donation was not really accounted ALL: There was no decision of what to do with the sponsorship as the board has no treasurer and we are working towards creating a foundation that can accept legally fundings. There was no ask for some amount of money for the devel course, no costs presented, no other details given. Nobody expressed a disagreement of giving funds for the devel course. BIA: 9th : replacement for Cesc as quick solution to deal with TM => suggests Henning Westerholt, all the other agree up to 17th DCM: asks Henning and then sends short announcement to community BIA: complains that he actually hasn't voted and the replacement was made in hurry (?!?!) DCM: presents that all other announcements were done more or less in the same fashion in the past, asking to propose improvements DCM: 24th - asks decision about local_route, board discusses and agrees to enclose in #ifdef HW: 25th - announces the result and asks BIA if he agrees, no response, he was away HW: after 1 day since asking BIA for agreement SVN is updated enclosing local_route in ifdefs BIA: 27th - disagrees with board decision, considering it was not the business of the board (sent to public as well) -- Note from DCM: opensips was registered at June 25, 2008, 17:30 according to http://sourceforge.net/projects/opensips/ Discussions with name proposals and foundation 2008-07 Others: discussions related to name and foundation HW: proposes a project renaming plan HW: on the 16th, vote for name: proposes Kamailio, majority reached on the 17th, last that agree (with a mention that he does not really like the name) BIA, on the 28th, after a call from HW where he mentioned he was away. AG: 17th - registers kamailio.org, ... AG: prposes unification of 1.4. and renaming => decided to announce first to allow time for renaming without pressure AG: 28th - announces renaming to public HW: some drafts of foundation if in Germany AG: details about alternative in Netherlands 2008-08 DCM: 4th - discovers news on voip-info.org that openser becomes opensips, Dan Pascu sends a mixed recipient message and some others see something is going on DCM: after short investigation and learning that behind opensips is bogdan decide to take some precaution actions and send details about them on board to decide if they still remain or roll back BIA: comes later on the mailing lists and announces the fork From then on most of the things were reflected on the public. No one was kicked out from the project, as developer, board member or simply list user, just some administrative rights were limited. Of course last three months have a lot of messages for the topics above. I hope I haven't missed something important and I ask other board members to correct or improve. With this one I hope things are a bit more clear about what board did so far, where were mistakes and wrong decisions. You judge! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] ser 2.1 - compilation error - conflicting types
by Tomasz Zieleniewski 08 Aug '08

08 Aug '08

Hi, When I try to compile ser from sources I get the following conflicting types errors: gcc -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev23-make"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.1"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lock_ops.c -o lock_ops.o In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:46: error: conflicting types for 'loff_t' /usr/include/linux/types.h:30: error: previous declaration of 'loff_t' was here /usr/include/sys/types.h:62: error: conflicting types for 'dev_t' /usr/include/linux/types.h:13: error: previous declaration of 'dev_t' was here In file included from /usr/include/sys/types.h:133, from ut.h:53, from lock_ops.c:28: /usr/include/time.h:105: error: conflicting types for 'timer_t' /usr/include/linux/types.h:22: error: previous declaration of 'timer_t' was here In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:198: error: conflicting types for 'int64_t' /usr/include/linux/types.h:98: error: previous declaration of 'int64_t' was here /usr/include/sys/types.h:204: error: conflicting types for 'u_int64_t' /usr/include/linux/types.h:97: error: previous declaration of 'u_int64_t' was here In file included from /usr/include/sys/types.h:220, from ut.h:53, from lock_ops.c:28: /usr/include/sys/select.h:78: error: conflicting types for 'fd_set' /usr/include/linux/types.h:12: error: previous declaration of 'fd_set' was here In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:235: error: conflicting types for 'blkcnt_t' /usr/include/linux/types.h:124: error: previous declaration of 'blkcnt_t' was here make: *** [lock_ops.o] Błąd 1 Is there any fix for that? Regards Tomasz

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[Kamailio-Users] mailing lists issues
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

Hello, I am not going in hot discussions here. We have to finish the release first. Due to DNS change of openser.org, without prior notice, we were facing about 1 hour down time with mailing lists. I apologize for that. Kamailio project should be reachable on two domains: - main one: kamailio.org (www.kamailio.org) - backup: kamailio.net (www.kamailio.net) The mailing lists domains are: - main one: lists.kamailio.org - backup: lists.kamailio.net If you have any problem, please contact me. Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] carrierroute compile error
by Mouncif Benniane 07 Aug '08

07 Aug '08

That fixed it thanks On Thu, Aug 7, 2008 at 4:03 PM, TCB <tawandac(a)gmail.com> wrote: > you need libconfuse-deve ! > > > On Thu, Aug 7, 2008 at 9:38 PM, Mouncif Benniane <mouncifbb(a)gmail.com>wrote: > >> I did install libconfuse rpm for centos 5 but still same compile results >> >> [root@r sip-server]# rpm -qa | grep confuse >> *libconfuse-2.5-4.el5* >> [root@r sip-server]# >> >> On Thu, Aug 7, 2008 at 3:30 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> You need libconfuse, don't know exactly the name of the package on >>> centos. >>> >>> http://www.kamailio.net/docs/modules/1.3.x/carrierroute.html#AEN53 >>> >>> Cheers, >>> Daniel >>> >>> PS. Please note that the new mailing list address is >>> users(a)lists.kamailio.net >>> >>> >>> On 08/07/08 22:25, Mouncif Benniane wrote: >>> >>>> I am compiling version openser 1.3.2 on centos realease 5 >>>> >>>> getting compile error stooping at carrierroute module see below: >>>> >>>> >>>> make[1]: Entering directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> make[1]: Leaving directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> make[1]: Entering directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> Compiling carrierroute.c >>>> Compiling carrier_tree.c >>>> Compiling load_data.c >>>> Compiling route_config.c >>>> route_config.c:39:21: error: confuse.h: No such file or directory >>>> route_config.c:50: error: expected declaration specifiers or â...â >>>> before âFILEâ >>>> route_config.c:52: error: expected â=â, â,â, â;â, âasmâ or >>>> â__attribute__â before â*â token >>>> route_config.c:63: error: expected â)â before â*â token >>>> route_config.c: In function âload_configâ: >>>> route_config.c:81: error: âcfg_tâ undeclared (first use in this >>>> function) >>>> route_config.c:81: error: (Each undeclared identifier is reported only >>>> once >>>> route_config.c:81: error: for each function it appears in.) >>>> route_config.c:81: error: âcfgâ undeclared (first use in this function) >>>> route_config.c:83: error: âdâ undeclared (first use in this function) >>>> route_config.c:83: error: âpâ undeclared (first use in this function) >>>> route_config.c:83: warning: left-hand operand of comma expression has no >>>> effect >>>> route_config.c:83: error: âtâ undeclared (first use in this function) >>>> route_config.c:83: warning: left-hand operand of comma expression has no >>>> effect >>>> route_config.c:93: warning: implicit declaration of function >>>> âparse_configâ >>>> route_config.c:104: warning: implicit declaration of function âcfg_sizeâ >>>> route_config.c:112: warning: implicit declaration of function >>>> âcfg_getnsecâ >>>> route_config.c:113: warning: implicit declaration of function >>>> âcfg_titleâ >>>> route_config.c:113: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:118: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:123: warning: implicit declaration of function >>>> âcfg_getintâ >>>> route_config.c:127: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:132: warning: implicit declaration of function >>>> âcfg_getfloatâ >>>> route_config.c:134: warning: implicit declaration of function >>>> âcfg_getstrâ >>>> route_config.c:134: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:135: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:137: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:145: warning: implicit declaration of function >>>> âcfg_getnintâ >>>> route_config.c:169: warning: implicit declaration of function âcfg_freeâ >>>> route_config.c: At top level: >>>> route_config.c:178: error: expected â=â, â,â, â;â, âasmâ or >>>> â__attribute__â before â*â token >>>> route_config.c: In function âsave_configâ: >>>> route_config.c:234: error: âFILEâ undeclared (first use in this >>>> function) >>>> route_config.c:234: error: âoutfileâ undeclared (first use in this >>>> function) >>>> route_config.c:241: warning: implicit declaration of function âfopenâ >>>> route_config.c:249: warning: implicit declaration of function âfprintfâ >>>> route_config.c:249: warning: incompatible implicit declaration of >>>> built-in function âfprintfâ >>>> route_config.c:251: error: too many arguments to function >>>> âsave_route_data_recursorâ >>>> route_config.c:257: warning: implicit declaration of function âfcloseâ >>>> route_config.c: At top level: >>>> route_config.c:274: error: expected declaration specifiers or â...â >>>> before âFILEâ >>>> route_config.c: In function âsave_route_data_recursorâ: >>>> route_config.c:280: warning: incompatible implicit declaration of >>>> built-in function âfprintfâ >>>> route_config.c:280: error: âoutfileâ undeclared (first use in this >>>> function) >>>> route_config.c:326: error: too many arguments to function >>>> âsave_route_data_recursorâ >>>> route_config.c: In function âbackup_configâ: >>>> route_config.c:335: error: âFILEâ undeclared (first use in this >>>> function) >>>> route_config.c:335: error: âfromâ undeclared (first use in this >>>> function) >>>> route_config.c:335: error: âtoâ undeclared (first use in this function) >>>> route_config.c:335: warning: left-hand operand of comma expression has >>>> no effect >>>> route_config.c:364: warning: implicit declaration of function âfeofâ >>>> route_config.c:365: warning: implicit declaration of function âfgetcâ >>>> route_config.c:366: warning: implicit declaration of function âferrorâ >>>> route_config.c:370: warning: implicit declaration of function âfputcâ >>>> route_config.c:377: error: âEOFâ undeclared (first use in this function) >>>> make[1]: *** [route_config.o] Error 1 >>>> make[1]: Leaving directory >>>> `/usr/local/src/openser-1.3.0/sip-server/modules/carrierroute' >>>> make: *** [modules] Error 2 >>>> >>>> >>>> >>>> >>>> any ideas? thank you >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> -- >>> Daniel-Constantin Mierla >>> http://www.asipto.com >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> > > > -- > TC >

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[Kamailio-Users] Accounting / REFER question
by Zahid Mehmood 07 Aug '08

07 Aug '08

Hi All, I have a need to keep the openser 'IN' the loop for all transfered calls and to be able to generate proper accounting records (to mimic legacy PBX behavior). Any thoughts on how this can be achieved WITHOUT using SBC/ B2BUA? is it even possible? If off-system user calls X on openser and then X transfers the call to Y, we need to be able to generate two cdr records: A --> X start when A called, end when Y disconnected X --> Y start when X called Y, end when Y disconnected. A is connecting to X via a cisco gateway. Y can potentially be another off-system number. Thanks in Advance. -- Zahid

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[Kamailio-Users] carrierroute compile error
by Mouncif Benniane 07 Aug '08

07 Aug '08

I am compiling version openser 1.3.2 on centos realease 5 getting compile error stooping at carrierroute module see below: make[1]: Entering directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' make[1]: Leaving directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' make[1]: Entering directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' Compiling carrierroute.c Compiling carrier_tree.c Compiling load_data.c Compiling route_config.c route_config.c:39:21: error: confuse.h: No such file or directory route_config.c:50: error: expected declaration specifiers or â...â before âFILEâ route_config.c:52: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token route_config.c:63: error: expected â)â before â*â token route_config.c: In function âload_configâ: route_config.c:81: error: âcfg_tâ undeclared (first use in this function) route_config.c:81: error: (Each undeclared identifier is reported only once route_config.c:81: error: for each function it appears in.) route_config.c:81: error: âcfgâ undeclared (first use in this function) route_config.c:83: error: âdâ undeclared (first use in this function) route_config.c:83: error: âpâ undeclared (first use in this function) route_config.c:83: warning: left-hand operand of comma expression has no effect route_config.c:83: error: âtâ undeclared (first use in this function) route_config.c:83: warning: left-hand operand of comma expression has no effect route_config.c:93: warning: implicit declaration of function âparse_configâ route_config.c:104: warning: implicit declaration of function âcfg_sizeâ route_config.c:112: warning: implicit declaration of function âcfg_getnsecâ route_config.c:113: warning: implicit declaration of function âcfg_titleâ route_config.c:113: warning: assignment makes pointer from integer without a cast route_config.c:118: warning: assignment makes pointer from integer without a cast route_config.c:123: warning: implicit declaration of function âcfg_getintâ route_config.c:127: warning: assignment makes pointer from integer without a cast route_config.c:132: warning: implicit declaration of function âcfg_getfloatâ route_config.c:134: warning: implicit declaration of function âcfg_getstrâ route_config.c:134: warning: assignment makes pointer from integer without a cast route_config.c:135: warning: assignment makes pointer from integer without a cast route_config.c:137: warning: assignment makes pointer from integer without a cast route_config.c:145: warning: implicit declaration of function âcfg_getnintâ route_config.c:169: warning: implicit declaration of function âcfg_freeâ route_config.c: At top level: route_config.c:178: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token route_config.c: In function âsave_configâ: route_config.c:234: error: âFILEâ undeclared (first use in this function) route_config.c:234: error: âoutfileâ undeclared (first use in this function) route_config.c:241: warning: implicit declaration of function âfopenâ route_config.c:249: warning: implicit declaration of function âfprintfâ route_config.c:249: warning: incompatible implicit declaration of built-in function âfprintfâ route_config.c:251: error: too many arguments to function âsave_route_data_recursorâ route_config.c:257: warning: implicit declaration of function âfcloseâ route_config.c: At top level: route_config.c:274: error: expected declaration specifiers or â...â before âFILEâ route_config.c: In function âsave_route_data_recursorâ: route_config.c:280: warning: incompatible implicit declaration of built-in function âfprintfâ route_config.c:280: error: âoutfileâ undeclared (first use in this function) route_config.c:326: error: too many arguments to function âsave_route_data_recursorâ route_config.c: In function âbackup_configâ: route_config.c:335: error: âFILEâ undeclared (first use in this function) route_config.c:335: error: âfromâ undeclared (first use in this function) route_config.c:335: error: âtoâ undeclared (first use in this function) route_config.c:335: warning: left-hand operand of comma expression has no effect route_config.c:364: warning: implicit declaration of function âfeofâ route_config.c:365: warning: implicit declaration of function âfgetcâ route_config.c:366: warning: implicit declaration of function âferrorâ route_config.c:370: warning: implicit declaration of function âfputcâ route_config.c:377: error: âEOFâ undeclared (first use in this function) make[1]: *** [route_config.o] Error 1 make[1]: Leaving directory `/usr/local/src/openser-1.3.0/sip-server/modules/carrierroute' make: *** [modules] Error 2 any ideas? thank you

2 3

[Kamailio-Users] test
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

please ignore -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] SIP 304 Temporary moved: sendto returning errno 1
by Vincent Pelletier 07 Aug '08

07 Aug '08

Hi. I built & installed ser-2.0.0-rc1 on a debian stable amd64. When I try to redirect a call (case where destination user redirects before answering), it gets an SIP error 477. Corresponding error from SER process running in foreground: 0(30330) ERROR: udp_send: sendto(sock,0x2aebd16fc038,1227,0,xx.xx.xx.xx:xxxx,16): Operation not permitted(1) 0(30330) msg_send: ERROR: udp_send failed 0(30330) ERROR: t_send_branch: sending request on branch 0 failed 0(30330) ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Unfortunately, "1" (EPERM) is not documented in sento manpage, as far as I can see, so it's quite hard to track down that problem. Kernel version is: 2.6.24.5-grsec iptables is used, but putting netcat in place of SER and the client shows that kernel allows packets to reach client (and more simply, it is possible to do a direct call, it's just redirects which fail). It can be a grsec problem, I still have to check that part, but I wanted to take the chance of helpful feedback in the meantime. Regards, -- Vincent Pelletier

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Re: [Kamailio-Users] www.openser.org
by Henning Westerholt 07 Aug '08

07 Aug '08

On Thursday 07 August 2008, Nuno Marques wrote: > Can someone tell me how to access the dokuwiki page? Is offline in both > projects (Kamailio and Opensips...). Hi Nuno, there is a wrong link in the page, should be fixed now. Cheers, Henning

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[Kamailio-Users] www.openser.org
by Bogdan-Andrei Iancu 07 Aug '08

07 Aug '08

Staring from today, www.openser.org will point to www.opensips.org . The mainlists will continue on openser like they use to, but I invite everyone to move to the new mail lists created on opensips.org 1 - It is the time to move forward with the openser project, in a positive way, for a reliable future development and enhancement. 2 - Kamillio community used the access right they had on the www.openser.org server to redirect to kamillio without asking my consent even if the domain is a property of Voice System. Thanks, Bogdan

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[Kamailio-Users] [Fwd: [Kamailio-Devel] branching 1.4.0 in one hour]
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

I think this is also useful info for user community as well. Cheers, Daniel -------- Original Message -------- Subject: [Kamailio-Devel] branching 1.4.0 in one hour Date: Thu, 07 Aug 2008 18:09:35 +0300 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: miconda(a)gmail.com To: devel <devel(a)lists.kamailio.org> Hello, this is a last notification between branching the 1.4.0 release. If any developers is still doing last minute work, let us know. Branching is about one hour from now 18:00 CET. Of course, will still be possible to add afterwards, just that will need commits in two branches. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Devel mailing list Devel(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Is sh interface is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi, I am working on OpenSER 1.3.2 version as presence server and i want to integrate with IMS. For this what are modifications need. Is sh interface is needed? And how to trace the presence information? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

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[Kamailio-Users] REFER & accounting question
by Zahid Mehmood 07 Aug '08

07 Aug '08

Hi All, I have a need to keep the openser 'IN' the loop for all transfered calls and to be able to generate proper accounting records (to mimic legacy PBX behavior). Any thoughts on how this can be achieved WITHOUT using SBC/ B2BUA? is it even possible? If off-system user calls X on openser and then X transfers the call to Y, we need to be able to generate two cdr records: A --> X start when A called, end when Y disconnected X --> Y start when X called Y, end when Y disconnected. A is connecting to X via a cisco gateway. Y can potentially be another off-system number. Thanks in Advance. -- Zahid

1 0

[Kamailio-Users] For xcap-client to make run is xcap-server is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi All, Presently i am working on OpenSER 1.3.2 version to make as presence server and i include the presence module. But i am getting some errors when i explored it seems xcap client was not include.Now i was include but my doubt is for xcap-client to make run is xcap-server is needed. If so where i can get it. Please help me on this. Thanks & Regards, Mahesh Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

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[Kamailio-Users] For xcap-client to make run is xcap-server is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi All, Presently i am working on OpenSER 1.3.2 version to make as presence server and i include the presence module. But i am getting some errors when i explored it seems xcap client was not include.Now i was include but my doubt is for xcap-client to make run is xcap-server is needed. If so where i can get it. Please help me on this. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] website updates
by Elena-Ramona Modroiu 07 Aug '08

07 Aug '08

Hi all, I have updated the website to reflect the renaming of the project. If you find broken links, please contact me. Best regards, Ramona

1 0

[Kamailio-Users] includes in the route script ?
by Francois Berganz 07 Aug '08

07 Aug '08

is it possible to divide the script route in others files and do includes? With that, I want to have one file for route[1] , one for route[2] .

2 1

Re: [Kamailio-Users] Users Digest, Vol 39, Issue 27
by luzango mfupe 07 Aug '08

07 Aug '08

Hi guys ..tired of this conflict..dont we supposed to have a mojor release today?? On 8/7/08, users-request(a)lists.kamailio.org <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: Question about registrar lookup() and branching. > (Gustavo Mistrinelli) > 2. Re: [Kamailio-Devel] [OpenSER-Users] OpenSER Advanced > Programming Course, Bucharest, Voice System, August 11-14, 2008 > (Andhika Gannesha Gemilang) > 3. Problem using additional attributes through radius_extra > parameter (Juha Heinanen) > 4. Re: [Kamailio-Devel] call for contribution: project logo > (Benny Prijono) > 5. Re: [Kamailio-Devel] call for contribution: project logo > (Torben Friese) > 6. Re: OpenSIPS project (Francois Menard) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 6 Aug 2008 09:59:43 -0300 > From: "Gustavo Mistrinelli" <gmistrinelli(a)gmail.com> > Subject: Re: [Kamailio-Users] Question about registrar lookup() and > branching. > To: "Alex Balashov" <abalashov(a)evaristesys.com> > Cc: users(a)lists.kamailio.org > Message-ID: > <c6717cc80808060559k24a7c774v96049c8fb5c7677a(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Alex, > > I don't know your complete architecture, but let me ask you why Asterisk is > not a trusted gateway? > The you can use rewritehost or LCR logic to route DIDs to Asterisk > If you need to associate users with DIDs, you can use dbaliases. > > Cheers, > > Gustavo > On Tue, Aug 5, 2008 at 8:55 PM, Alex Balashov > <abalashov(a)evaristesys.com>wrote: > >> Greetings, >> >> I've got a small problem with how registration lookups work. >> >> I have several concurrent registrations from one PBX that have multiple >> contact URIs that they declare that aren't the same; they contain the >> DIDs to route a call to. So, for instance, in Asterisk: >> >> register => user:pass@registrar/DID1 >> register => user:pass@registrar/DID2 >> >> Of course, lookup() doesn't care what those contacts actually are. If >> it sees multiple contacts for one username, t_relay() will simply create >> branches for all of them and and ring them all. I can't actually have >> that happen. >> >> I'm looking for suggestions on how to handle this most elegantly. The >> two options I've got in mind are: >> >> 1. Stop using lookup() and do my own URI rewrites and branch forking (if >> the contact URIs are identical) with attention to the contact. >> >> 2. Use BRANCH-ROUTEs to arrest the call leg if it is going to an RURI >> user part that does not match the user part of the contact URI, and... >> drop() the branches? >> >> I also need to build in some other logic, such that for example if a >> user has only one contact registered and it does not match the DID, or >> none of the contacts match the DID, proceed as normal anyway. >> >> Is it possible there is a better way? Recommendations welcome. >> >> Cheers, >> >> -- Alex >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

3 3

[Serusers] call via several SER server
by 静于 07 Aug '08

07 Aug '08

hello all, Could two UAs call each other via several SER proxy? I want to implement this function.How to do it?Do I need to modify the ser.cfg? How dose SER server connect each other? --------------------------------- 雅虎邮箱，您的终生邮箱！

1 0

[Kamailio-Users] Problem using additional attributes through radius_extra parameter
by Ruchir 06 Aug '08

06 Aug '08

Hi, I'm trying to configure openser & freeradius with AAA. I'm using custom mysql tables for authentication & accounting. I've modified queries in sql.conf in raddb folder. Authentication works fine. I've added additional fields in radius_extra parameter of acc module. I've added those parameters in dictionaries of radiusclient-ng & freeradius. But I get blank values for custom fields. Any idea what can be wrong? *openser.cfg* modparam("acc", "radius_extra","User-Name=$fU;Dialstatus=$avp(s:dialstatus);User-Rate=$avp(i:2016);User-Duration=$avp(i:2016);User-Clid=$fU;User_ClidName=$avp(s:callerid);User_Dst_Name=$avp(s:2014);User_Dst_Code=$avp(i:2015);User_Account_Id=$fU;Lcr_Rate=$avp(i:2030);Lcr_Duration=$avp(i:2016);Lcrtrunk_Id=$avp(i:2027);Lcr_Dst_Name=$avp(s:2028);Lcr_Dst_Code=$avp(i:2029);Real_Duration=$avp(i:2016)") *dictionary.openser* ATTRIBUTE Dialstatus 1073 string ATTRIBUTE User-Rate 1074 string ATTRIBUTE User-Duration 1075 string ATTRIBUTE User-Channel 1076 string ATTRIBUTE User-Clid 1077 string ATTRIBUTE User_ClidName 1080 string ATTRIBUTE User_Dst_Name 1081 string ATTRIBUTE User_Dst_Code 1082 string ATTRIBUTE User_Account_Id 1083 string ATTRIBUTE Lcr_Rate 1084 string ATTRIBUTE Lcr_Duration 1085 string ATTRIBUTE Lcrtrunk_Id 1086 string ATTRIBUTE Lcr_Dst_Name 1087 string ATTRIBUTE Lcr_Dst_Code 1088 string ATTRIBUTE DialedTime 1089 string ATTRIBUTE Real_Duration 1090 string ATTRIBUTE Startleg 1091 string ATTRIBUTE Stopleg 1092 string *dictionary.radius* ATTRIBUTE Dialstatus 1073 string ATTRIBUTE User-Rate 1074 string ATTRIBUTE User-Duration 1075 string ATTRIBUTE User-Channel 1076 string ATTRIBUTE User-Clid 1077 string ATTRIBUTE User_ClidName 1080 string ATTRIBUTE User_Dst_Name 1081 string ATTRIBUTE User_Dst_Code 1082 string ATTRIBUTE User_Account_Id 1083 string ATTRIBUTE Lcr_Rate 1084 string ATTRIBUTE Lcr_Duration 1085 string ATTRIBUTE Lcrtrunk_Id 1086 string ATTRIBUTE Lcr_Dst_Name 1087 string ATTRIBUTE Lcr_Dst_Code 1088 string ATTRIBUTE DialedTime 1089 string ATTRIBUTE Real_Duration 1090 string ATTRIBUTE Startleg 1091 string ATTRIBUTE Stopleg 1092 string

2 1

[OpenSER-Users] OpenSER Advanced Programming Course, Bucharest, Voice System, August 11-14, 2008
by Bogdan-Andrei Iancu 06 Aug '08

06 Aug '08

Hi everybody, Following the tradition started last year, the second edition of the "OpenSER Advanced Programming Course" will be held in Bucharest, Romania, starting with 11th of August 2008. The primary goal is the same: to attract more developers to OpenSER and build a higher development squad. There are three major upgrades from last edition: - free participation - there will be no fees for attending to this course; you just need to register. - there will be 4 days course instead of 3 days - the course will contain also a practical work where the students will have the chance to consolidate the knowledge they have accumulated during the theoretical presentations. The course will have a 60% - 40% distribution between theory and practice. The course is dedicated to people willing to learn how to develop OpenSER extensions. The course will follow the idea of writing a new openser module from scratch, with all the dependencies it has from the core: dynamic memory usage, MI interface, registering timers, locking, exporting pseudo-variables, etc The course is sponsored courtesy of : - Packt Publishing (donating 5% of the revenue from the Flavio E. Goncalves's "Building Telephony Systems with OpenSER" book.) - Voice System (logistics and hosting) Teachers from the OpenSER development team will be: - Bogdan-Andrei Iancu - Anca Vamanu As, said, the registration is free, but the class is limited as size up to 17 - 20 people. The seats will be granted on the "first in, first served" policy :). For registration, you need to mail me (bogdan(a)voice-system.ro) or to openser(a)voice-system.ro . You will receive a confirmation response to guarantee your registration. Best regards, Bogdan

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[Serusers] Textops problem
by Frank Hase 06 Aug '08

06 Aug '08

Hello, I´m using the SER for my phones at home. Now I want to phone as an anonymous. So I add a line in the ser.cfg: subst('/^From:(.*)sip[^@]*@[a- zA-Z0-9.]+(.*)$From:anonymous@anonymous.invalid;/ig'); But this doesn´t work. I just want to have in the Message Header the line From: anonymous(a)ananymous.invalid; So that I can phone anonym. Thank you for your help. Greeting Frank Haase

2 1

Re: [Serusers] Serusers Digest, Vol 64, Issue 4
by Akintayo Olusegun 06 Aug '08

06 Aug '08

mail it again On Wed, Aug 6, 2008 at 11:00 AM, <serusers-request(a)lists.iptel.org> wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. 2.0 survey (Jiri Kuthan) > 2. ser 2.1 - compilation error - conflicting types > (Tomasz Zieleniewski) > 3. ser 2.1 - compilation error - conflicting types > (Tomasz Zieleniewski) > 4. SER/FIFO not accessible (cindy.cao) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 05 Aug 2008 19:20:33 +0200 > From: Jiri Kuthan <jiri(a)iptel.org> > Subject: [Serusers] 2.0 survey > To: "serusers(a)iptel.org" <serusers(a)iptel.org> > Message-ID: <48988BE1.9030503(a)iptel.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I'm just wondering if folks using the 2.0 have found so far any issue > which has not been tracked -- I've been off-line for vacation, IETF and > a couple of other trips and began to be uncertain if I might have missed > something. Any concerns/issues/comments regarding 2.0 are most welcome, > publicly or privately. > > Thanks a lot > > -jiri > > > ------------------------------ > > Message: 2 > Date: Wed, 6 Aug 2008 01:11:34 +0200 > From: "Tomasz Zieleniewski" <tzieleniewski(a)gmail.com> > Subject: [Serusers] ser 2.1 - compilation error - conflicting types > To: serusers(a)lists.iptel.org, serusers(a)iptel.org > Message-ID: > <5fd52d7a0808051611g2017ff27uef0685ef960450d(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-2" > > Hi, > > When I try to compile ser from sources I get the following conflicting > types > errors: > gcc -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops > -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' > -DVERSION='"2.1.0-dev23-make"' -DARCH='"x86_64"' -DOS='linux_' > -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.1"' -D__CPU_x86_64 -D__OS_linux > -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' > -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM > -DSHM_MMAP > -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE > -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST > -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK > -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM > -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD > -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM > -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT > -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lock_ops.c -o > lock_ops.o > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:46: error: conflicting types for 'loff_t' > /usr/include/linux/types.h:30: error: previous declaration of 'loff_t' was > here > /usr/include/sys/types.h:62: error: conflicting types for 'dev_t' > /usr/include/linux/types.h:13: error: previous declaration of 'dev_t' was > here > In file included from /usr/include/sys/types.h:133, > from ut.h:53, > from lock_ops.c:28: > /usr/include/time.h:105: error: conflicting types for 'timer_t' > /usr/include/linux/types.h:22: error: previous declaration of 'timer_t' was > here > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:198: error: conflicting types for 'int64_t' > /usr/include/linux/types.h:98: error: previous declaration of 'int64_t' was > here > /usr/include/sys/types.h:204: error: conflicting types for 'u_int64_t' > /usr/include/linux/types.h:97: error: previous declaration of 'u_int64_t' > was here > In file included from /usr/include/sys/types.h:220, > from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/select.h:78: error: conflicting types for 'fd_set' > /usr/include/linux/types.h:12: error: previous declaration of 'fd_set' was > here > In file included from ut.h:53, > from lock_ops.c:28: > /usr/include/sys/types.h:235: error: conflicting types for 'blkcnt_t' > /usr/include/linux/types.h:124: error: previous declaration of 'blkcnt_t' > was here > make: *** [lock_ops.o] B??d 1 > > Is there any fix for that? > > Regards > Tomasz >

1 0

[Kamailio-Users] Question about registrar lookup() and branching.
by Alex Balashov 06 Aug '08

06 Aug '08

Greetings, I've got a small problem with how registration lookups work. I have several concurrent registrations from one PBX that have multiple contact URIs that they declare that aren't the same; they contain the DIDs to route a call to. So, for instance, in Asterisk: register => user:pass@registrar/DID1 register => user:pass@registrar/DID2 Of course, lookup() doesn't care what those contacts actually are. If it sees multiple contacts for one username, t_relay() will simply create branches for all of them and and ring them all. I can't actually have that happen. I'm looking for suggestions on how to handle this most elegantly. The two options I've got in mind are: 1. Stop using lookup() and do my own URI rewrites and branch forking (if the contact URIs are identical) with attention to the contact. 2. Use BRANCH-ROUTEs to arrest the call leg if it is going to an RURI user part that does not match the user part of the contact URI, and... drop() the branches? I also need to build in some other logic, such that for example if a user has only one contact registered and it does not match the DID, or none of the contacts match the DID, proceed as normal anyway. Is it possible there is a better way? Recommendations welcome. Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 10

[Kamailio-Users] Authentication problems.
by Buzukelis 06 Aug '08

06 Aug '08

*Hi all, ** I'm trying to register few ata's to openser.* *i puted two registrations to subscribers table.* *but registaration allwais unauthorized.* ** ** *ngrep output: * *interface: eth0 (192.168.27.0/255.255.255.0) filter: (ip) and ( port 5060 )* *U 192.168.27.117:10186 -> 192.168.27.201:5060 REGISTER sip:192.168.27.201 SIP/2.0. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK645148049;rport. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2000 REGISTER. Contact: *. Max-Forwards: 70. User-Agent: Grandstream HT-502 V1.0B 1.0.1.8. Expires: 0. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Length: 0. .* * U 192.168.27.201:5060 -> 192.168.27.117:10186 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK645148049;rport=10186. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b963. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2000 REGISTER. WWW-Authenticate: Digest realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.27.117:10186 -> 192.168.27.201:5060 REGISTER sip:192.168.27.201 SIP/2.0. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK142179605;rport. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2001 REGISTER. Contact: *. Authorization: Digest username="1000", realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7", uri="sip:19$ Max-Forwards: 70. User-Agent: Grandstream HT-502 V1.0B 1.0.1.8. Expires: 0. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Length: 0. . U 192.168.27.201:5060 -> 192.168.27.117:10186 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.27.117:10186;branch=z9hG4bK142179605;rport=10186. From: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201>>;tag=604945105. To: <sip:1000@192.168.27.201 <sip%3A1000(a)192.168.27.201> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.3c26. Call-ID: 53873712-10186-1(a)192.168.27.117. CSeq: 2001 REGISTER. WWW-Authenticate: Digest realm="192.168.27.201", nonce="48998a3ba4544026eab5a8f7cf93dc258a5d26d7". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. openser.cfg :# # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # ------------------ module loading ---------------------------------- #set module path mpath="//lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest. if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; save("location"); exit; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; }; consume_credentials(); lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } can not understand why this configuration is not working to me??? any help will be apreciated. *

1 0

[Kamailio-Users] Change the INVITE timer after append_branch()
by Jens Carl 06 Aug '08

06 Aug '08

Hello List, I have call forwarding on no answer and the possibility to set the time out for the call. I do this by setting the value of "fr_inv_timer" with "avp_fr_timer_avp". Then when the call is route to user's phone and he has call forwarding on no answer activate I overwrite the default INVITE timer with his value. If the call is answered with the error message "Request Timeout (408)" I replace the RURI with the new number, call append_branch() and relay the message to myself. Now the message goes again to my routing logic and will be relayed. I also set a new timeout for the INVITE message, but still the value of the customer is used. For example if the users has a ring time of 10 seconds the second call also rings only for 10 seconds. How can I set the ringtime after append_branch() to my default value again? Is this possible or should I change me routing logic to handle this kind of feature? When I have to change it what is the best way to do this? Thanks in advance Jens

2 2

[Kamailio-Users] 1.4.0 release plans, project matters
by Henning Westerholt 06 Aug '08

06 Aug '08

Hi all, Bogdan-Andrei Iancu prepared this fork from Kamailio (formerly known as OpenSER) since July. This was done secretly, we were not aware of this. We've all done a great amount of work to resolve the existing trademark issues and create a better organisational backing for the project. Many contributors tried hard to improve the quality of the code base, a big number of bug and documentation fixes were done, doxygen code documentation was greatly extended. Because of the project renaming our release schedule needed some adaptions. We discussed this in the last week on the developer list and found a new release date for Kamailio [1]. Version 1.4.0 will be released on the thursday, the 7. august. Cheers, Henning Westerholt [1] http://lists.kamailio.org/pipermail/devel/2008-July/014926.html

9 21

[Serusers] SER/FIFO not accessible
by cindy.cao 06 Aug '08

06 Aug '08

2008-08-06 cindy.cao >Hello all: > >I am trying to add a user to mysql database with the command: > serctl add 1000 1000 1000(a)ser.com > but I get an error: SER/FIFO not accessible! >and i consulted some documents which said it should be the permission >problem. so i insert a line "fif0_mode=0777" into my "ser.cfg" and try the >command again, but now my system halt! > > Any pointers to where to start debuging this error? > Thanks so much > > Best Regards. >Cindy

1 0

[Kamailio-Users] call for contribution: project logo
by Daniel-Constantin Mierla 05 Aug '08

05 Aug '08

Hello, I already received the first proposal, courtesy of Darren Sessions. You can check at: http://www.kamailio.org/pub/kamailio-logos/ This message is an invitation for all of you to submit proposals for the new project logo. We are aware that the name is not making happy many people out there and we can work to solve it, maybe over the time will not look that bad or just send name suggestions. However, this should not stop you being creative and help the project, in a transitional or definitive state. Looking forward to your contributions. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] 2.0 survey
by Jiri Kuthan 05 Aug '08

05 Aug '08

I'm just wondering if folks using the 2.0 have found so far any issue which has not been tracked -- I've been off-line for vacation, IETF and a couple of other trips and began to be uncertain if I might have missed something. Any concerns/issues/comments regarding 2.0 are most welcome, publicly or privately. Thanks a lot -jiri

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[Kamailio-Users] Openser version to run Mediaproxy 2.0
by Nuno Marques 05 Aug '08

05 Aug '08

Hello, I was reading the README file on the download page of Mediaproxy and found this statement: This version of MediaProxy (2.0 or higher) cannot be used in combination with any version of OpenSER older than 1.4 or any components of MediaProxy older than 2.0. You must completely upgrade any previous installation of OpenSER and MediaProxy to these new versions. What i want to ask you is if this is really true because Openser 1.4 (kamailio) isn't available yet... Can i use Mediaproxy 2.0 with Openser 1.3.2 or i need to wait for the Openser 1.4? Thanks in advance, Nuno Marques

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[Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008
by Daniel-Constantin Mierla 05 Aug '08

05 Aug '08

Hello Everybody, as many expressed their intention for an openser/kamailio event in Bucharest, next week, I am going to organize a dinner in a place with nice Romanian beer, wine and food. Seeing the suggestions for the date, I am proposing August 11, 2008 (Monday) -- let me know if you have opinions. Exact hour and place will be sent as soon as I get an idea about number of participants. So far, expressed the intention to participate: - Henning Westerholt, developer and board member - Elena-Ramona Modroiu, developer and board member - Daniel-Constantin Mierla, developer and board member - Michiel ten Hagen - Sebastian Schuman Please write me if you want to attend, I need to know the number of attendants to make the reservation. The event is in the same fashion as the ones happening in Barcelona, Vienna, Berlin, ..., in the past -- everybody pays for its expenses. If you want to have some preamble (might not be same place, but...), you can check some photos from last similar event in Bucharest after the ROSDEV2008 conference: http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/index1.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] CDRTool: no price showing
by Brian Del Shasta 05 Aug '08

05 Aug '08

David, After double checking, I do seem to be getting E.164 formatted numbers coming from the Cisco datasource, at least for the numbers we care about. Following is an example, you can see that I am able to successfully match a 1800 # as 1800, test (I just can't rate a price to it). id,StartTime,Username,CallerParty,Destination,DestinationId,DestinationName,Duration,Price,Gateway,KBIn,KBOut,Disconnect,Channel,Direction,CallType 1,2008-08-05 08:55:28,,5635833713,2415453,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/2:D:2,outgoing,Telephony 2,2008-08-05 08:55:28,,5635833713,2415453,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/0:D:1,incoming,Telephony 3,2008-08-05 08:55:11,,6082516000,19206750230,,,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/1:D:22,outgoing,Telephony 4,2008-08-05 08:55:11,,6082516000,19206750230,,,0,,,0,0,0,Normal release,,incoming,VoIP 5,2008-08-05 08:54:37,,6082888583,18004266918,1800,test,0,,208.66.128.72-ISDN-7,0,0,0,Normal release,ISDN 7/1:D:18,outgoing,Telephony On Fri, Aug 1, 2008 at 8:23 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > correct > > On Fri, Aug 1, 2008 at 8:11 PM, Brian Del Shasta <briandelshasta(a)gmail.com > > wrote: > >> Okay, I'm using a Cisco datasource, so I'm assuming that would replace >> OpenSER in the general schema, correct? >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

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[Kamailio-Users] Problem With SipTrace Module
by Ali Jawad 05 Aug '08

05 Aug '08

Hi I did enable the siptrace module to be able to see SIP packets in TLS mode. Once I did set it up, I could not logon to the server anymore. The log file does not show anything : Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=acc Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=nathelper Aug 5 10:28:21 [30781] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=siptrace Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: connection type is xxxxxxx via TCP/IP Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: protocol version is 10 Aug 5 10:28:21 [30793] DBG:mysql:db_mysql_new_connection: server version is 5.0.48 Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=xlog Aug 5 10:28:21 [30793] DBG:xlog:child_init: init_child [-4] pid [30793] Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=acc Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=nathelper Aug 5 10:28:21 [30793] DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4, module=siptrace And the server is listening correctly. # ----------------- SIP Trace Info modparam("siptrace", "db_url", "mysql://mysql://openser:xxxxxxxxx@xxxxxxxxxxxx/openser") modparam("siptrace", "trace_on", 1) Regards

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[OpenSER-Users] XMPP/SIP-SIMPLE Problem
by Cesar Troncoso 05 Aug '08

05 Aug '08

Hi everyone, I have a problem with xmpp module. Messages going from xmpp to sip-simple work fine. The problem I have is in the opposite direction (from sip-sipmle to xmpp). If I set fork=no or children=1 in my config file then all messages are sent and received in both directions. This is what I see in the logs: xmpp: invalid content-type 0x300ff xmpp: unable to write to command pipe: Bad file descriptor Any Idea how to overcome this situation? Thanks! Cesar

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