sr-users September 2012

sr-users@lists.kamailio.org

78 participants
138 discussions

AUTOREPLY call forking using dbaliases not working for...

by strobel＠hfo-telecom.de

******************* AUTOREPLY VON yufei.tao(a)redembedded.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 17.09.2012 bis einschließlich 21.09.2012 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Künzel oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hi I have a strange problem on forking calls to a group of users. For example I have two users y2 and y3 in dbaliases, both with alias_username 'group'. And y2 and y3 both registers with Kamailio fine. When I make a call to 'group' from a third client y1, what my kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to BRANCH_ALIASDB, where a lookup location will be done for each of the username resulting from lookup of dbaliases, something like this: y1--INVITE 'group'-->lookup dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup location---¬ | |-->relay ---->[BRANCH_ALIASDB]--->'y3'-->lookup location---- The all works well as long as all clients are NAT'ed. However when they are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only goes to one of the group members, e.g. y2 only. When checking the log, it seemed to have done the dbaliases lookup fine, and each location lookup successfully. But Kamailio only relayed y2's IP, e.g. to the client, while y3's to itself. When comparing the location table when clients are NAT'ed or not, I find that the 'received' column is only populated when I do fix_nated_register. And group calls only works when 'received' column is populated. That explains why when clients are NAT'ed group calls work, as I only do fix_nated_register if nat_uac_test returns true. But if this is the only reason, if two clients register using the same username, e.g. both as y3, and when 'received' column of location table is empty (no fix_nated_register done), I would expect a call to y3 should also only make 1 client ring. But in fact both of them rang! The flow is like: y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client registered as 'y3' | ---> IP of 2nd client registered as 'y3' While a call to 'group' (thus dbaliases lookup took place) under such un-NAT'ed set up made only 1 client ring. So I can make it work by always doing fix_nated_register. But I'm not clear about these things: - why does a lookup of dbaliases before lookup of location make such difference? - does lookup location work differently depending on whether it is called from trunk or from a route called from a branch route? Following is relevant parts from my config file: ############################################################# route[LOCATION] { if ( alias_db_lookup("dbaliases") ) { t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB], # call another route that looks up location, # if not existent, call drop() } else { xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n"); route(LOCATION_TRUNK); # normal look up location and sending of 404 etc } ... ... } ############################################################# branch_route[BRANCH_ALIASDB] { xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n"); route(LOCATION_BRANCH); } route[LOCATION_BRANCH] { if (!lookup("location")) { # Drop this branch - it's going nowhere drop(); } } ############################################################# route[RELAY] { xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ############################################################ route[NAT] { #!ifdef WITH_NAT xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); force_rport(); if (nat_uac_test("2")) { if (method=="REGISTER") { fix_nated_register(); xlog("L_DBG","NAT: Just done fix_nated_register in REGISTER message in NAT route\n"); } else { xlog("L_DBG","NAT: fix_nated_contact\n"); fix_nated_contact(); } setflag(FLT_NATS); } # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force media relay #!endif return; } ############################################################# And in the main route, route LOCATION and RELAY are the last two routes: route { ... route(NAT); ... ... # user location service route(LOCATION); route(RELAY); } Hope I have made it clear. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

11 years, 11 months

record_route_advertised_address question

by Juha Heinanen

rr readme on record_route_advertised_address function tells: Note: A second Record-Route will be inserted if the transport used on the inbound and outbound interfaces changes. that is not the case. i made call record_route_advertised_address("192.98.103.10:5060"); when request came over udp and went out over tcp and only one rr header was inserted: incoming: U 2012/09/17 17:30:17.467079 192.98.103.10:5060 -> 192.98.103.2:5070 INVITE sip:test@test.fi SIP/2.0. ... outgoing: T 2012/09/17 17:30:17.504631 192.98.103.10:5060 -> 192.98.103.10:38078 [AP] INVITE sip:0x20b7580@192.98.103.10:5050;transport=tcp SIP/2.0. Record-Route: <sip:192.98.103.10:5060;ftag=wumod;lr>. ... i have not set enable_double_rr param and default is 1. is this a bug that i should report on the tracker? -- juha

11 years, 11 months

Port change 5060

by Vijay Thakur

All Experts, I want to change my SIP port from 5060 for other one. Before making any change in my live server (Kamailio 3.2.1, i want to be sure. Kindly suggest me that where should i make changes to implement it. This is a security measure for kamailio from port scanning. With thanks in advance. Vijay Thakur

11 years, 11 months

pidf-lo extraction

by Cenk İlker İzanlı

Hi, Is there any basic document showing how to extract info from pidf-lo data on kamailio part? Thx.

11 years, 11 months

rsyslog settings for good kamailio performance

by Albert Petit

Hi, For load testing I would like to setup rsyslog so Kamailio logs are written asynchronously Would the following configuration be enough for that purpose in * rsyslog.conf*? # # Write log file in asynchronous mode and use an IO Buffer of 64 Kb # IO buffer config http://www.rsyslog.com/doc/rsyslog_conf_global.html # # $OMFileAsyncWriting on $OMFileFlushOnTXEnd off $OMFileFlushInterval 60 $OMFileIOBufferSize 128k #---------------------------------------------------------------------- ## kamailio ##---------------------------------------------------------------------- $outchannel kamailio_log_rotation,/var/log/pocserverplatform/kamailio.log,524288000,/usr/local/bin/kamailio_log_rotation_script :programname, contains, "kamailio" $kamailio_log_rotation local0.* $kamailio_log_rotation Apart of that is there a way to completely disable completely any logs of Kamailio to discard they're causing any performance issue? Thanks Albert

11 years, 11 months

kamailio prepaid services

by Cenk İlker İzanlı

Hi, Is there any guide to install/configure kamailio as prepaid service infrastructure? Thx

11 years, 11 months

Re: [SR-Users] Limiting concurrent calls

by Alex Balashov

Htable is another popular method for those disposed less favourably to the dialog module. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Uri Shacked <ushacked(a)gmail.com> wrote:in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

11 years, 11 months

Limiting concurrent calls

by Uri Shacked

in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri > Hi, > I need to limit the number of concurrent calls go through kamailio. > Its a simple set up(Not per user basis, a global limit will suffice). > > I know i need to employ dialog module. But i'm a little confused about > profiles. I thought for simple application like this, i may be able to > use profile altogether. > > Any code snippet to limit number of concurrent calls will be very helpful. > > Thanks in advance. > -- > -aft

11 years, 11 months

I get no calls?

by gooly

Hi, I have registered my (Fritz!Box) ATA successfully registered on my account 8305234(a)iptl.org but what is tried I never received a call? Here is what I can see on 'my' web-page with you: > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:58 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:56 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:20 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:19 500 Server Internal Error Please can you tell me what I have to do to be able to receive Voip-Calls? Regards, Carl

11 years, 11 months

Limiting concurrent calls

by Aft nix

Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

11 years, 11 months

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sr-users September 2012