sr-users June 2015

sr-users@lists.kamailio.org

82 participants
135 discussions

Append to reply in default action of failure_route

by Efelin Novak

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

8 years, 6 months

kamailio 4.0.1 websocket configuration

by Rupayan Dutta

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

8 years, 6 months

Kamailio - Asterisk, problem with SIP trunk mode

by Mickael MONSIEUR

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

8 years, 7 months

Issue with SNMPstats

by Igor Potjevlesch

Hello, I have configured the SNMPstats module fine in my machine because I get the SNMP information with the new KAMAILIO MIBS but I don't understand why the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 However I can see the transactions: KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32: 54797 Should I configure an specific parametr of the SNMPstats module? Regards,

8 years, 7 months

Kamailio IMS - VM

by Bruno Cruz

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

8 years, 8 months

Header field Contact not changed on kamailio 4.2

by Зуев Михаил

Hi all, I need to modify the host part of a contact header. I'm trying something like: route { ..... if (remove_hf("Contact")){ if (insert_hf("Contact: rn", "Contact")) xlog("Contact modified"); } if (!t_relay("YYY.YYY.YYY.YYY","5060")) { sl_reply_error(); } ...... } When i look SIP DUMP on YYY.YYY.YYY.YYY side i see older Contact field. I try subst_hf with regexp, subst and simply remove Contact, all this does not work. What i am doing wrong? -- С уважением Зуев Михаил

8 years, 8 months

problem with dispatcher load balancing

by Ding Ma

I'm trying to set up kamailio dispatcher to distribute calls to 2 asterisk servers. So far, the failover case seems ok, but I cannot get the dispatcher to distribute load. All calls are going to the last destination entry in the dispatcher table even if I have set the maxload attributes. I'm using algorithm 10 for load distribution. The number of calls sent to one asterisk is well above the maxload. Hope someone more experienced in dispatcher can review my config and give me some recommendations. Thanks in advance. Here is the content of dispatcher table in postgresql db. select * from dispatcher; id | setid | destination | flags | priority | attrs | description ----+-------+--------------------+-------+----------+---------------------------+------------- 1 | 1 | sip:10.0.1.31:5061 | 0 | 0 | duid=asterisk1;maxload=25 | Asterisk1 2 | 1 | sip:10.0.1.33:5061 | 0 | 0 | duid=asterisk2;maxload=25 | Asterisk2 (2 rows) Here are the dispatcher sections in kamailio.cfg. # ----- dispatcher params ----- #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", DBASTURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "force_dst", 1). # If flag 2 is set, then failover support is enabled. modparam("dispatcher", "flags", 3) # the last address in destination set is used as a final option to send the request to modparam("dispatcher", "use_default", 1) # load balancing fail over modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # PVs for hashing modparam("dispatcher", "hash_pvar", "$fU@$ci") # PVs to store results when calling ds_is_from_list modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") # method to probe the gateways modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:dispatcher@localhost") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) # size of hash table storing data for call load dispatching, power of two modparam("dispatcher", "ds_hash_size", 10) # expiration time in seconds to remove the load on a destination if no BYE was received modparam("dispatcher", "ds_hash_expire", 3600) # expiration time in seconds to remove the load on a destination if no 200 OK for INVITE was received # and state updated with ds_load_update modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 30) #!endif route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_DISPATCHER if(is_method("BYE|CANCEL") && ds_is_from_list("1", "3")) ds_load_update(); #!endif ...... } } route[FROMASTERISK] { #!ifdef WITH_DISPATCHER if(ds_is_from_list("1", "3")) { xlog("L_DBG","$rm from $fU@$si:$sp: Call from Asterisk cluster\n"); return 1; } return -1; #!else if ($si==$sel(cfg_get.asterisk.bindip)) { return 1; } return -1; #!endif } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); #!ifdef WITH_DISPATCHER if(is_method("INVITE") && ds_is_from_list("1", "3")) { if(status=~"2[0-9][0-9]") { ds_load_update(); } else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_DISPATCHER failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { # mark the destination Inactive and Probing ds_mark_dst("IP"); # select the new destination if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } else { # last available node failed to reply, no other destinations available send_reply("404", "No destination"); exit; } } } #!endif route[TOASTERISK] { #!ifdef WITH_DISPATCHER # ds_mark_dst("IP"); # Call load distribution if(!ds_select_dst("1", "10")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","$rm from $fU@$si:$sp: No destinations available for $rd\n"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); #!else $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); #!endif route(RELAY); exit; } # Dispatcher detects a destination goes down event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } # Dispatcher detects a destination comes up event_route[dispatcher:dst-up] { xlog("L_ERR", "Destination up: $rm $ru\n"); }

8 years, 9 months

A question to you, SIP providers: what is your policy on PBXes calling themselves

by Antonio Gómez Soto

Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two options: 1. The PBX notices this, and directly connects the phone to the DID on that system (breaking separation of tenants) 2. The PBX sends the call out on the SIP trunk, and the provider-routing sends the call back as an incoming call. What are the pros and cons of each option from the SIP provider point of view? How do PBX's generally behave? Thanks, Antonio PS: I reposted, because the original question was apparently not phrased in a clear way.

8 years, 10 months

IPSec supporting open source SIP Server

by Priyaranjan Nayak

Hi All, I need one IPSec supported opensource SIP server for testing my sip client endpoint.Could you please tell me which SIP server supporting IPSec ? Thanks in advance. Thanks Priyaranjan

8 years, 10 months

Announcement: Kamailio is now systemd-rtc-server

by Alex Balashov

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina,was quick to summarise the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

8 years, 11 months

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sr-users June 2015