http://permalink.gmane.org/gmane.comp.voip.ser hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://permalink.gmane.org/gmane.comp.voip.ser/49968 <pre>Hello, I recently started using SCTP relay feature of Kamailio and am receiving SCTP-SIP and  relaying it UDP-SIP and vice versa. What happens , if  an SCTP association is broken from the distant end in the middle of the call, is there a way to re transmit the SCTP message via another route ? Thanks, --Jignesh Gandhi _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users&lt; at &gt;lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </pre> Jignesh Gandhi 2013-05-20T04:12:08 http://permalink.gmane.org/gmane.comp.voip.ser/49967 <pre>Hi Barry: This issue have not been resolved after following by your method that modifed the video1_sipregs table struct,attachment is table info and asterisk log. Can you help me with this problem? thank you very much! best zhengyw *asterisk log* asterisk_log.txt &lt;http://sip-router.1086192.n5.nabble.com/file/n118567/asterisk_log.txt&gt; *dbinfo.txt* dbinfo.txt &lt;http://sip-router.1086192.n5.nabble.com/file/n118567/dbinfo.txt&gt; -- View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-x-and-Asterisk-10-7-0-Realtime-Integration-tp118248p118567.html Sent from the Users mailing list archive at Nabble.com. </pre> zhengyw 2013-05-20T03:07:33 http://permalink.gmane.org/gmane.comp.voip.ser/49966 <pre>i'm trying to use the example kamailio.cfg to route to voicemail server on busy or decline. Only thing I did was adding decline code to t_check_status("486|408"), enabling the preprocessor variable for voicemail and changing the voicemail host and port to my voicemail server. No requests arrive on my voicemail server and the dial tone keeps ringing even if phone is busy and when I decline on the receiving phone there's a new INVITE sent to the phone directly afterwards. Am I doing something wrong here? </pre> hiro 2013-05-19T12:05:01 http://permalink.gmane.org/gmane.comp.voip.ser/49965 <pre>http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc#dokuwiki__top Hello I was trying to use your tutorial on the above page. I am trying to do so using PGQL vs MYSQL. I changed all the mysql parts to pgsql however when i try to start kamailio it fails. I am building the system on alpine linux and attached my kamailio.cfg and kamctlrc. i am able to setup using the default config making necessary changes but when i use yours it fails. Any help would be appreciated. </pre> Carlton Thompson 2013-05-18T19:10:35 http://permalink.gmane.org/gmane.comp.voip.ser/49964 <pre>Hi, I have a problem when using uri_db module. When I leave everything on default and not specify modparam "uri_db", "db_url", then it will not connect to the database. In the var/log/syslog I have seen that it tries to connect to database okmaialio instead of kamailio. Is it bug, or did I misconfig something? I left the kamctlrc file intact and everything to default... can you confirm this please? Thank you! _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users&lt; at &gt;lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </pre> Ján Hrnko 2013-05-18T20:26:43 http://permalink.gmane.org/gmane.comp.voip.ser/49963 <pre>Hi there, i hope you can help me, i am having trouble with the register page of siremis, strange thing is, i can open the register page, but only after i have logged my self in, if not, it redirects me directly to the login page.. how can i setup the register page to be accessible without the need of logging in ? Also i am getting the info after trying to register, that "Public registration is not enabled!" but i have set siremis/modules/ser/config/common.Main.php to $cfg_siremis_public_registrations = true; What is the problem, please help. Greetings, Alex </pre> Alexander Albert 2013-05-18T01:22:15 http://permalink.gmane.org/gmane.comp.voip.ser/49962 <pre>Hello, I'm using a very ugly code to test remove_body(): request_route { remove_body(); $rd = "192.168.254.85"; t_relay(); } Original INVITE (with SDP): --------------------------------- U 192.168.254.102:5060 -&gt; 192.168.254.104:5060 INVITE sip:123&lt; at &gt;192.168.254.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK50cdeeb8. Max-Forwards: 70. From: "10005" &lt;sip:10005&lt; at &gt;192.168.254.102&gt;;tag=as78ddcd16. To: &lt;sip:123&lt; at &gt;192.168.254.104&gt;. Contact: &lt;sip:10005&lt; at &gt;192.168.254.102:5060&gt;. Call-ID: 79a1c78f01914f1814bfcb7d622cf735&lt; at &gt;192.168.254.102:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.7.1. Date: Sat, 18 May 2013 05:47:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 269. . v=0. o=root 1763381182 1763381182 IN IP4 192.168.254.102. s=Asterisk PBX 1.8.7.1. c=IN IP4 192.168.254.102. t=0 0. m=audio 16162 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-even</pre> Konstantin M. 2013-05-18T06:00:59 http://permalink.gmane.org/gmane.comp.voip.ser/49961 <pre>Hi Daniel We need to modify Kamailio in a proper way to have separate connection per call. We need your guidance and we will share the details so that it will be useful for somebody else. Now how do we proceed. My thought is 1. In Kamailio, there might be a structure or object that might be storing per call information. Let's call it *call_context*. Also there might be another data structure that might be storing registration information.Let's call it *reg_context*. 2. In tcp_main.c , tcp_do_connect() creates socket . So I want to store tcp_connection structure in call_context. So that there are separate sockets per call. 3. Same thing should be done for registration, like storing tcp_connection structure in per reg_context. Now my question is it the right approach, or kindly advise what is the right approach. If what I have mentioned is right approach, pls give some information regarding data structures and how can I store specific tcp_connection structure in a call_context or reg_context. We are tota</pre> Kamal Palei 2013-05-18T04:58:42 http://permalink.gmane.org/gmane.comp.voip.ser/49960 <pre>Hello, I want to announce that a new person got developer GIT write access to repository: Victor Seva. He helped pushing back Kamailio in official Debian packages and lately has submitted many patches to various modules, among upcoming ones is some work on newly added per module debug level (see the logs from yesterday's IRC meeting for more about this feature). His git commit id is: vseva My warm welcome and looking forward to future work within the project! Cheers, Daniel </pre> Daniel-Constantin Mierla 2013-05-17T14:31:58 http://permalink.gmane.org/gmane.comp.voip.ser/49959 <pre>Yeah, nice Jobs ! And it works ! :p Alexis -----Message d'origine----- De : sr-users-bounces&lt; at &gt;lists.sip-router.org [mailto:sr-users-bounces&lt; at &gt;lists.sip-router.org] De la part de Olle E. Johansson Envoyé : vendredi 17 mai 2013 16:15 À : Kamailio (SER) - Users Mailing List Objet : [SR-Users] IMS In Kamailio 4.0.0 - what is it? Jason Penton, Richard Good and Carsten Bock has written an excellent article about the IMS extensions in Kamailio 4.0.0. What it is, what you can do with it and what they're currently working on. Go read it on our web site now to learn more about this work: http://www.kamailio.org/w/2013/05/ims-kamailio/ Have a great weekend! /O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users&lt; at &gt;lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _________________________________________________________________________________________________________________________ Ce message et ses piec</pre> alexis.marcou< at >orange.com 2013-05-17T14:21:31 http://permalink.gmane.org/gmane.comp.voip.ser/49958 <pre>Hello, Daniel-Constantin! OK, thank you. I think need example and more detailed description. See attach, but i think need corrections. -- WBR, Victor JID: coyote&lt; at &gt;bks.tv JID: coyote&lt; at &gt;bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users&lt; at &gt;lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </pre> Victor V. Kustov 2013-05-17T13:46:39 http://permalink.gmane.org/gmane.comp.voip.ser/49957 <pre>Jason Penton, Richard Good and Carsten Bock has written an excellent article about the IMS extensions in Kamailio 4.0.0. What it is, what you can do with it and what they're currently working on. Go read it on our web site now to learn more about this work: http://www.kamailio.org/w/2013/05/ims-kamailio/ Have a great weekend! /O </pre> Olle E. Johansson 2013-05-17T14:14:40 http://permalink.gmane.org/gmane.comp.voip.ser/49956 <pre> i quickly looked the code and seems like rule_id would need to be added to matched_gw_info struct when load_gws() is called and copied from there as a new field to gw_uri_avp. then when next_gw() is called, rule_id stored with matched gw would need to be copied from the gw's matched_gw_info struct to a new rule_id_avp. </pre> Juha Heinanen 2013-05-17T14:02:00 http://permalink.gmane.org/gmane.comp.voip.ser/49955 <pre>Hello, if you send a patch, we will review and commit appropriately. There should be a way to detect is ipv4 or ipv6, because at least the size is different. In kamailio AVPs it should be stored as string ... Cheers, Daniel On 5/17/13 2:46 PM, Victor V. Kustov wrote: </pre> Daniel-Constantin Mierla 2013-05-17T13:31:45 http://permalink.gmane.org/gmane.comp.voip.ser/49954 <pre>Dear All, We would like to store SIP text Messages when the destination Subscriber is Offline.We have insert into kamailio cfg file the configuration lines below.Unfortunately storing messages is unsuccessful.Any ideas of what missing or what could be wrong? Best regards. ************************************************************************************************************loadmodule "msilo.so" #!ifdef WITH_MSILOmodparam("msilo","db_url","mysql://[% kamailio.proxy.dbrwuser %]:[% kamailio.proxy.dbrwpw %]&lt; at &gt;[% database.dbhost %]/[% kamailio.proxy.dbname %]")modparam("msilo", "db_table", "silo")modparam("msilo","from_address","sip:registrar&lt; at &gt;xxxxxxx.local")modparam("msilo", "from_address", "sip:$rU&lt; at &gt;xxxxxxx.local")modparam("msilo","contact_hdr","Contact: &lt;sip:registrar&lt; at &gt;xx.xx.xx.xx:5062&gt;;msilo=yes\r\n")modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n")modparam("msilo","offline_message","*** User $rU is offline!")#!endif modparam("usrloc", "db_mode", 0) initial value was “1” ################</pre> andreas tseiko 2013-05-17T13:22:04 http://permalink.gmane.org/gmane.comp.voip.ser/49953 <pre>Hi I did setup kamailio to handle websocket, and I am testing using pjsip, my problem is that I can logon using the pjsip client but SIP does not register, pjsip displays a SIP connection timeout error. My xlite client can logon just fine though. The log from kamailio can be seen below http://pastebin.com/XrsQM8Cf Please advice _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users&lt; at &gt;lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </pre> Ali Jawad 2013-05-17T12:51:59 http://permalink.gmane.org/gmane.comp.voip.ser/49952 <pre>Hello, Daniel-Constantin! Consider http://www.rfc-editor.org/errata_search.php?rfc=3162 --VERIFIER NOTES-- The common terminology is known by RADIUS implementers. Every RADIUS implementor "knows what this means". i.e. "Address" in RFC 2865 means "ipaddr", and "Address" in RFC 3162 means "ipv6addr". As we see it different types. In current parsing misc_radius just skip all attributes except string, integer, ipaddr with warning ("attribute %d is not of type integer"). I think need complete support for all attributes types with typecast in avp's: integer and string. Maybe part of typecasts will useful in common transformations. -- WBR, Victor JID: coyote&lt; at &gt;bks.tv JID: coyote&lt; at &gt;bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux </pre> Victor V. Kustov 2013-05-17T12:46:49 http://permalink.gmane.org/gmane.comp.voip.ser/49951 <pre> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Adding the SIP SRV records did the trick. I can register and make calls with jitsi using TLS. I have also tested calling a user on my Kamailio server from a user registered on this one and vice versa. Both calls were successful so federation works. Thanks again for setting this up. I'd love to see a step by step wiki on how you did this. Even if it is a rough first draft. Getting a working system up and running for the first time is quite a task. Cheers, John On 16/05/13 19:35, Daniel-Constantin Mierla wrote: have time for it soon, so I enabled back srv records. I will add naptr soon. done. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRlhhFAAoJELy1jPQ1KER7eoIP/jVDfhQwUf4bT3VM5OdvYmyg zCcMBkUcPmVAWgU7pJYkF9dANSzI2/U81xK3d2UxaT+Jlty6PLskjVA1Bje9ouAA XuaUApefzMl/nGVJNgovQkrcl4H+1M4H5lbQCKkYsuNp2oAzlYBNfsoTvbbpQuB7 HmiixpdXAscRVAbrqnhi2hmerm5ceNh63TZhBOaUoCV</pre> johnc 2013-05-17T11:45:09 http://permalink.gmane.org/gmane.comp.voip.ser/49950 <pre>Hi Thanks Juha for your response. Any solution to check the rule extra info. I want route billing info to put in the acc. Any standar solution? Cheers On Fri, May 17, 2013 at 11:27 AM, Juha Heinanen &lt;jh&lt; at &gt;tutpro.com&gt; wrote: _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users&lt; at &gt;lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users </pre> Eloy Coto Pereiro 2013-05-17T10:43:41 http://permalink.gmane.org/gmane.comp.voip.ser/49949 <pre> no, that is not currently available in any avp. </pre> Juha Heinanen 2013-05-17T10:27:41 http://permalink.gmane.org/gmane.comp.voip.ser/49948 <pre> I tried video call, using two Android devices running Linphone with VP8 codec, and it worked ok. Make sure your both ends meet in at least one supported codec (e.g. Linphone on Linux does not support VP8, while Linphone on Android does not support H263, MPV4, theora, x-snow..) With regards Pavel </pre> Pavel Kasparek 2013-05-17T09:00:21 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.voip.ser