July 2015 - sr-users - kamailio.org

Kamailio mod_destroy() ERROR
by Ali Taher 22 Jul '15

22 Jul '15

Hello, I'm getting below error when starting Kamailio. ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Any advice ? Regards, Ali

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replace From header username for AUTH
by Al S 22 Jul '15

22 Jul '15

Hi, I was wondering how can I replace From header username to perform the following authentication check with the new value: route[AUTH]......if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "1"); exit;}... Thanks,Al

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kamailio 4.3v partition column name is reserved word in mysq ndbcluster engine
by José Seabra 21 Jul '15

21 Jul '15

Hello Daniel, After update my kamailio to version 4.3 and change the table location as required in migration process, kamailio start giving some errors: [root@ccp1 kamailio]# 35(7637) ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'partition' in 'where clause' 35(7637) ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query that's is because the column name "partition" is reserved word ( http://dev.mysql.com/doc/refman/5.6/en/keywords.html) in mysql ndbcluster engine, so what i did to fix this issue was add a new modparam in modules/usrloc/ul_mod.c file: {"partition_col", PARAM_STR, &partition_col}, with this i can configure what name i want for this column in kamailio module configuration and configure a diferent column name in table location(ex: modparam("usrloc", "partition_col", "partitionusrloc") ), Can you please check out if what i did is the correct way to fix it, in order to avoid damage something else. Best regards -- Cumprimentos José Seabra

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Compiling only one module
by Loic Chabert 21 Jul '15

21 Jul '15

Hello, I'm writing a new module for influxdb stats. I've wrote my code and now i need to compile it. How can i compile it ? I don't want to compile all modules, only this one. Thanks in advance, Regards.

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usage of t_flush_flags() in event_route
by Andrew Pogrebennyk 20 Jul '15

20 Jul '15

Hi Daniel and others, I'm having a problem with acc module if I'm using the event_route/ branch-failure: say, the call comes from the app server and goes to the registered user. We arm the the failure route and per-branch failure route for the 302 redirect from the UA. We explicitly reset the accounting flags because we don't want to account the calls from the app server. The transaction is created implicitly by the t_relay(). Now if the UA responds with the 302 response and we are going to process that, we want to create an acc record for the new target from 302 message because this call may incur additional costs. I'm setting the accounting flags and calling t_flush_flags() but that doesn't work (no accounting record). Any idea if I'm doing something wrong or maybe there's a bug when changing the flags and then calling t_flush_flags from the event_route? Here are the module parameters: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "db_url", PAIR_URL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "cdr_log_enable", 0) FTR, we tried 4.1.6 and 4.1.8. And here is the event route (with flags defined like this: flags FLAG_ACC_DB:1, FLAG_ACC_MISSED:2, FLAG_ACC_FAILED:3, ...): event_route[tm:branch-failure:redirect] { route(ROUTE_STOP_RTPPROXY_BRANCH); if($T_rpl($rs) == 301 || $T_rpl($rs) == 302) { # initialise variables when entering failure route route(ROUTE_INITVARS); # these need to be avps because we need it in reply/failure-route $(avp(s:from_faxserver)[*]) = 0; $(avp(s:to_faxserver)[*]) = 0; $(avp(s:cf_from_pstn)[*]) = 0; $(avp(s:from_pstn)[*]) = 0; $(avp(s:proxylu_from_pstn)[*]) = 0; $(avp(s:lcr_flags)[*]) = 0; $(avp(s:em_call)[*]) = 0; $(avp(s:from_pbx)[*]) = 0; $(avp(s:p_to_device)[*]) = 0; $(avp(s:p_to_group)[*]) = 0; $(avp(s:is_primary)[*]) = 0; # now let's process a 30x $(avp(s:acc_state)[*]) = "cfb"; $(avp(s:orig_acc_caller_user)[*]) = $avp(s:acc_caller_user); $(avp(s:orig_acc_caller_domain)[*]) = $avp(s:acc_caller_domain); $(avp(s:acc_caller_user)[*]) = $avp(s:acc_callee_user); $(avp(s:acc_caller_domain)[*]) = $avp(s:acc_callee_domain); $(avp(s:caller_uuid)[*]) = $avp(s:callee_uuid); $(avp(s:callee_uuid)[*]) = $null; # the $var(no_acc) is 0 at this point but the flags may have been reset # if this is a call from PBX user - we do want accounting for the 302 redirect if(isflagset(FLAG_ACC_DB)) { xlog("L_NOTICE", "++++++ ACC flag is set - [% logreq -%]\n"); } else { xlog("L_NOTICE", "------ ACC flag is NOT set - [% logreq -%]\n"); } setflag(FLAG_ACC_FAILED); setflag(FLAG_ACC_DB); t_flush_flags(); # get last URI from destination-set and set it as R-URI $var(contact) = $T_rpl($ct); $var(contact) = $(var(contact){nameaddr.uri}); if($var(contact) == 0 || $var(contact) == $null) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [% logreq -%]\n"); acc_db_request("480", "acc"); $var(announce_handle) = "callee_tmp_unavailable"; $var(announce_set) = $xavp(callee_real_prefs[0]=>sound_set); $(avp(s:announce_code)[*]) = 480; $(avp(s:announce_reason)[*]) = "Temporarily Unavailable"; route(ROUTE_EARLY_REJECT); } $ru = $var(contact); xlog("L_NOTICE", "Redirect from UAC intercepted - [% logreq -%]\n"); $(avp(s:forwarder_cli_userprov)[*]) = $T_rpl($tU); $(avp(s:forwarder_domain_userprov)[*]) = $T_rpl($td); $var(forward) = 1; $var(redirected_forward) = 1; route(ROUTE_LOAD_CALLER_PREF); route(ROUTE_FIND_CALLEE); } } Thanks. Andrew

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File Transfer
by Abdul Hakeem 20 Jul '15

20 Jul '15

Hello, Is there a support for file transfer between registered SIP clients, If not, how can this be achieved ? Regards, Abdul Hakeem

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Kamailio v4.3.1 Released
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, Kamailio SIP Server v4.3.1 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.3.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.1. For more details about version 4.3.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Missing BYE at the far end
by Kliment Toshkov, Netfinity JSC 20 Jul '15

20 Jul '15

Dear colleagues, We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE. Current setup is the following: PEER IP > KAMAILIO 4.2.2 > (Asterisk PBX 1, 2, 3, …, N) We provision only termination services, which is: 1) peer calls us 2) Kamailio chooses destination using dispatcher module 3) forwards the call to it. Attached is captured trace. Clearly BYE is retransmitted a few times, then request for UPDATE is received but it is not possible to answer it because the call is already hung up (as far as I understand). Kamailio routing file is available upon request. Thank you in advance! Климент Тошков Изпълнителен директор 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

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Planing to release v4.3.1
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, I am considering to release the first patch version in 4.3 series by beginning of next week (Monday or Tuesday, July 20 or 21). If you are aware of issues not reported yet, then open an item on the tracker to be taken care: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

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Double Record-Route to subscriber with multiple locations on different SIP transports
by Andrey Utkin 17 Jul '15

17 Jul '15

Hi! We found that we need double Record-Route when we call from one SIP transport to another (let's say, TCP -> TLS), because without |transport=" hint in Record-Route, mobile Linphone app sends messages (INVITE reply, ACK, BYE) via UDP, even if configured for another transport. record_route_advertised_address() is told to add double RR if needed, automatically, so we use it. But the subscriber (the receiver of call) can have multiple registered locations, let's say, one connected via TCP, and another via TLS. So we need to handle each call branch added by lookup(). But adding record_route_advertised_address() ONLY to branch_route attached by t_on_branch() doesn't execute if there's only one location. Having record_route_advertised_address() just in routing script after lookup() seems wrong because it seems unable to edit it for other branches (I don't remember if I have tested this and have proven a failure, or this is just my speculation). So we have added it in both places - right under lookup(), and in branch_route. But in this case we have an error logged: ERROR: rr [rr_mod.c:337]: w_record_route_advertised_address(): Double attempt to record-route Could anyone explain what would be the most correct solution? What is intended behaviour of branch_route? Is it omitted from execution on "primary" branch of call? Is there any parameter which could help? Thanks in advance. -- Andrey Utkin

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