sr-users May 2013

sr-users@lists.kamailio.org

95 participants
157 discussions

Radius - new style AVP's (misc_radius)
by Victor V. Kustov 17 May '13

17 May '13

Good day! Now I test new style AVP's for radius response storing. Today additional radius responses return in special radpair: SIP-AVP - name, string "email:sr-users@lists.sip-router.org session-timeout#161 nexthopip:h323/0001111@myvoip-gate.kamailio.org" - value. I don't know why developers chose this way. But it's addition kamailio's feature, so let it be. But if radius server responded with standard pairs? Sending Access-Accept of id 60 to 172.16.X.X port 59736 Session-Timeout = 4261674 next-hop-ip = "SIP/00111222333444@cisco-out" SIP-AVP = "email:sr-users@lists.sip-router.org session-timeout#161 next-hop-ip:h323/0001111@myvoip-gate.kamailio.org" session-protocol = "SIP" No chance to get full radius response at this moment. Only parse SIP-AVP pair. This mean that when you wanna integrate K. in existing infrastructure (in my case - billing), u need work with K. and radius. But why, if you already had worked voip devices? I think more flexible solution ADD in misc_radius function to get radpairs as AVP's. See that i got: misc_radius [functions.c:159]: AVP 'Session-Timeout'/0='null'/4261674 has been added misc_radius [functions.c:143]: AVP 'next-hop-ip'/0='SIP/005555777888@cisco-out'/0 has been added misc_radius [functions.c:143]: AVP 'session-protocol'/0='SIP'/0 has been added After radius response i get: $avp(Session-Timeout)=4261674 (integer) $avp(next-hop-ip)="SIP/005555777888@cisco-out" (string) $avp(session-protocol)="SIP" (string) I think its usable and more flexible. I see 3 way to integrate it: 1. additionally with existing radius_load_caller_avps/radius_load_callee_avps (i make same for test) 2. new function in same module, like radius_authorize_request (caller,callee). 3. replace existing style. it's bad for backward compability, of cause. Or it stay my local patch :) If it really needed and usable i may made any (1..3) way with doc and examples :) diff --git a/modules/misc_radius/functions.c b/modules/misc_radius/functions.c index 2a42024..f156939 100644 --- a/modules/misc_radius/functions.c +++ b/modules/misc_radius/functions.c @@ -107,6 +107,60 @@ error: return -1; } +static void generate_avps_rad(VALUE_PAIR* received) +{ + int_str name, val; + unsigned short flags; + VALUE_PAIR *vp; + + vp = received; + + for( ; vp ; vp=vp->next) { + flags = AVP_NAME_STR; + switch(vp->type) + { + case PW_TYPE_STRING: + flags |= AVP_VAL_STR; + name.s.len = strlen(vp->name); + val.s.len = strlen(vp->strvalue); + name.s.s = vp->name; + val.s.s = vp->strvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + case PW_TYPE_INTEGER: + name.s.len = strlen(vp->name); + name.s.s = vp->name; + val.n = vp->lvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + default: + LM_ERR("skip attribute type %d (non-string)", vp->type); + continue; + } + return; + } +} + /* Generate AVPs from Radius reply items */ static void generate_avps(struct attr *attrs, VALUE_PAIR* received) @@ -212,6 +266,7 @@ int radius_load_caller_avps(struct sip_msg* _m, char* _caller, char* _s2) LM_DBG("success\n"); rc_avpair_free(send); generate_avps(caller_attrs, received); + generate_avps_rad(received); rc_avpair_free(received); return 1; } else { -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

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Msilo configuration
by andreas tseiko 17 May '13

17 May '13

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 %]@[% database.dbhost %]/[% kamailio.proxy.dbname %]")modparam("msilo", "db_table", "silo")modparam("msilo","from_address","sip:registrar@xxxxxxx.local")modparam("msilo", "from_address", "sip:$rU@xxxxxxx.local")modparam("msilo","contact_hdr","Contact: <sip:registrar@xx.xx.xx.xx:5062>;msilo=yes\r\n")modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n")modparam("msilo","offline_message","*** User $rU is offline!")#!endif modparam("usrloc", "db_mode", 0) initial value was “1” #########################################################################Store messages to offline Subs########################################################################route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay();} failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); };}***********************************************************************************************

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Kamailio and PJSIP Issue
by Ali Jawad 17 May '13

17 May '13

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

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[OT] announcement: openrcs.com - free SIP service
by Daniel-Constantin Mierla 17 May '13

17 May '13

Hello, following several requests during Kamailio World, but also in the direction of federating open source and local communities via SIP (as a result of FOSDEM panel and discussions), a SIP service was built up to be offered for free on best effort manner - the hardware and bandwidth are offered by asipto.com, the service is using the domain openrcs.com. Main goal is to offer a free option for community members (but not restricted to) to create and use SIP accounts. The plan is to enable many (common) new features to allow people to play with, helping also to test kamailio. There is a rather basic (and alpha stage) web portal based on wordpress+buddypress, good enough for the moment to create an account and tune some options - it is available at: - https://www.openrcs.com Along with voice/video calling and instant messaging, the service has enabled features such as simple presence, msrp relay, xcap server and websocket (web portal embeds a bare sip app built using jssip library). Other features will be enable over the time, but have in mind **it is a not a telephony service**, thus no such specific services like emergency, pstn interconnect, etc. Hope it will be useful for many people - who ever wants to try it, head to https://www.openrcs.com ! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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config
by johnc 16 May '13

16 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Daniel, I nearly forgot, can you post your openrcs config ? Many thanks, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRlOvSAAoJELy1jPQ1KER7juoP/RmguFzHtfzdyr+RydqtE/xp TDfyzQCOh/iciln+pXANMtgajwyyHa83504HP05uhXnk3RLvLEoze/8t7/KkwSdR mksANtDJ/cR4Q9LI9iXVplIUORIR98qGl0/C+LODSU5Hsd3/FS5V0v0B+d1gEZPV na4UPlu/FGgCbRfUa/eFVfo/gE38fU8TNbtZqwggSYmmXrKdEUlzqN8cxCeFg0fT LVygThcLhtam1r4NQWpXOjzHl+qDrVNnIwuDng0W1kg3OLvt2GMS2l0FsLglZcMf xr6CcVQT+BvKUKGgubF1PfjcMuVhNGXgLH4dcXEbNs5Z41pDL0ap7nnz/SXiZQxa qHNEgFV9cJQJtPGK0fTuIPVV4YFA7iaMoXlLJbvwI5YwZUG9W3WnytDBuLUcXnSG 0E8Js9KHUHj975BQMaMUFDohLKL1kFv+wiRKbxT0QDVsEtMTMugi3vkLGRgskrNA e798aVReyh+8gB1hcm9bEUCt7Pc/aQBWYiLM40JqdlVS2Y28X4jQ8BYtSVROLGQa Ws0/nspwKWIXmC5DKF1AJNlBI1HSdW78PO1R2hc5uCatjcidthrvOmCwwhC3B/yA 9rPD0ECquOkRzLw7IHIFzM9pAruRDlKVdjhRaoN4hMuN57e/nqKuASPpMMXbj45L 4rSCPsSV9/A7kCLd9miV =88yR -----END PGP SIGNATURE-----

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Delay in "100 Trying" reply message
by Robert R 16 May '13

16 May '13

Hi, There is a big delay in our Kamailio server when repling "100 Trying" after incoming INVITE messages (more than 500 ms). It seems my perl script (and database query) is executed before "100 Trying" being send out. I was wondering if sending "100 Trying" is configurable so I can send it right after INVITE and before executing my perl script. request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); if (is_method("INVITE")) { perl_exec("test"); } t_check_trans(); route(SIPOUT); route(PRESENCE); route(REGISTRAR); .. route(RELAY); } (morthen Thanks R

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IRC Development Meeting
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, it's probably the time to have a IRC development meeting to set the targets for the next major release as well as discuss what is needed these days around the project. I proposed next week on Thursday, May 16, at 14:00GMT (16:00 Berlin time). I created a page to collect the topics, only few from my list being added for now: * http://www.kamailio.org/wiki/devel/irc-meetings/2013a Feel free to add topics there. Also, I would be available on Tuesday at the same time if proves to be more convenient for more people. If none of these dates suit for majority, we can look for other alternatives -- just propose and we can start a poll. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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event_route
by Bruno Bresciani 16 May '13

16 May '13

Hi All, in a call forking, after one branch answer the call (200 OK reply), a CANCEL SIP message has been sending to other/another branch(es) and I need to process this/these cancellations in configuration file. After reading some documentations, I discovered there is event_route[tm:local-request] block, which is executed when tm generates internally and sends a SIP request, Such cases are: SIP messages sent by msilo module SIP messages sent by presence server SIP messages sent by dialog module SIP messages sent via MI or CTL interfaces I didn't understand very well this cases, so I insert event_route block in my kamailio.cfg but neither CANCEL SIP message or other requests generated by tm module was handled by event_route. I must be using wrong concept to handle this CANCEL SIP message, it's possible handle this messages in configuration file? Best Regards

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redirecting irc channel #sip-router to #kamailio
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, already suggested on the irc channels, I plan to redirect #sip-router to #kamailio to group in a single place and have consistency with the name of the project. That means making #sip-router an invite-only channel and automatically redirecting those that want to join to #kamailio. Hope is ok with everyone, invite-only flag can be removed later. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1
by Trevor Burt 16 May '13

16 May '13

Dear All, using the following link:- http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I have successfully integrated Asterisk and Kamailio on the same box for testing, but am now facing the problem of getting Freepbx to use the same MySQL database tables. My Kamailio and Asterisk install uses the following tables:- sipusers sipregs voicemail voicemail_data voicemail_messages However Freepbx uses the following tables. devices users Any idea on how I fully integrate Freepbx with my Asterisk and Kamailio configuration using Mysql? My goal is to have a Kamailio LB to multiple Asterisk/Freepbx servers sharing the same extension and voicemail databases. I will also probably integrate a2billing and share the a2billing database between the Asterisk/Freepbx servers to create a fully scalable solution. Any advice most gratefully received.

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sr-users May 2013