February 2016 - sr-users

no pkg memory left
by Gholamreza Sabery 15 Feb '16

15 Feb '16

Hello. I am seeing the following error repeating again and again in the log file: /sbin/kamailio[14510]: ERROR: registrar [reply.c:199]: build_contact(): no pkg memory left I am using 64MB of SHM memory and 8MB of PKG memory. What is the reason for this error? How to solve it?

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kamailio 4 with asterisk
by Sébastien Brice 15 Feb '16

15 Feb '16

Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. the only thing i missed is asterisk behaviors'r regarding sip registration ? I tryed to place a call between 102 and 103 extensions and experimenting an issue Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail ! -- Executing [103@public:1] Dial("SIP/102-00000001", "SIP/103") in new stack [Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] apart that my sip.conf and extensions.conf are very minimal: [LocalSets] exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,102,Voicemail(${EXTEN},b) exten => _1XX,103,Hangup [general] context=LocalSets ; Default context for incoming calls. Defaults to 'default' rtcachefriends=yes ; Cache realtime friends by adding them to the internal list i did INSERT TO the users in mysql tables (sipusers, sipregs and voicemail) and registering extensions from UA works ok (i am using jitsi) Whats wrong with my setup ? thank you

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kamctl add VS kamctl ul add under SIPML
by kssyrr 15 Feb '16

15 Feb '16

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uac module configuration
by Andrew Tan 15 Feb '16

15 Feb '16

Hello, Submit a ticket in GitHub apologize about that I should have sent my question here. I would like some help in configuring the uac module to route to an gateway without specifically setting the URI-domain from the UAC. I would like to set change it in the Kamailio side. Currently I use uac_auth on failure route. All is working if I set the URI from on the client to xxx(a)gatewayaddress.com. But if I just dial just the number as is which translate to xxx(a)kamailioserver.com the invite doesnt go anywhere. My failure route is like so failure_route[MANAGE_FAILURE] { xlog("DEBUG: MANAGE FAILURE ROUTE: "); if ($T_reply_code == "407") # there's also 401 { $avp(arealm) = GATEWAY; $avp(auser) = USERNAME; $avp(apass) = PASSWORD; #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); #rewritehost("sip.ca.didlogic.net"); uac_auth(); rtpengine_offer("force trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"); route(NATMANAGE); route(RELAY); exit; } } Thank you

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LCR
by Alexey V. Panfilov 15 Feb '16

15 Feb '16

Hello, I'm using LCR module in my kamailio setup. I can access to the gw_uri_avp's value of matched gateway, but I'd like to get value of selected id of lcr_rule_target. Is there any way to get it? Thanks. P.S. Access to value of custom table's field (for example, named as 'flags') of cr_rule_target will be better. I'd like to store at this field the price of selected direction. -- Best regards, Alexey V. Panfilov

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Kamailio 4 && Asterisk. Getting started
by Sébastien Brice 14 Feb '16

14 Feb '16

Hi everyone I did follow the tutorial Kamailio 4.0.x and asterisk written by Daniel-C and I need a hand :-) after a successful install (kamailio and asterisk on the same server), i did setup 2 sip extensions (102 and 103) and tryed to place a call between them and experimenting an issue Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail ! -- Executing [103@public:1] Dial("SIP/101-00000001", "SIP/103") in new stack [Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] Actually none of my extensions are online and i am wondering if Kamailio forward register with asterisk ? transport between K and Asterisk seems ok (at least i did setup bindport in both sip.conf and kamailio.cfg) apart that my sip.conf and extensions.conf are very minimal: exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,102,Voicemail(${EXTEN},b) exten => _1XX,103,Hangup [general] context=LocalSets ; Default context for incoming calls. Defaults to 'default' rtcachefriends=yes ; Cache realtime friends by adding them to the internal list I am a bit new with Kamailio and I don't know if this behavior is normal since i have a sipregs mysql table that could do the job on purpose? Could someone point me to the right direction and enlight my knowing, thanks you thx you

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Kamailio and freeswitch integration for SBC
by malik sherif 12 Feb '16

12 Feb '16

Hello, I was able to have successful call with kamailio and after integrating kamailio with freeswitch using the following link, the invite timeout and as a result the call failed.The status on kamailio show these info. Do I need to download outbound model? I am running kamailio ver 4.1.1 and freeswitch 1.4.18. kamailio[19153]: 0(19162) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob kamailio[19153]: 0(19162) INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc Thank you very much and your help is appreciated. Thanks Abdul <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc [Asipto ...<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> kb.asipto.com The scope of this tutorial is to show how you can use Kamailio (former OpenSER) and FreeSWITCH to build a complete SIP/VoIP platform for large number of subscribers.

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BLF on different IPs/domains
by Phil Lavin 12 Feb '16

12 Feb '16

Hi all, A different BLF query to my other one so creating another thread. We've tried to have a BLF client SUBSCRIBE using a secondary IP on Kamailio. It is not sent a NOTIFY, but a client which has done a SUBSCRIBE using the primary IP is. If we route the problematic client such that it connects to the primary IP, it too is sent a NOTIFY. Looking at the query that is done on the database when a call is made to 441164969988 it is as follows: select `callid`,`to_tag`,`from_tag` from `active_watchers` where `presentity_uri`='sip:441164969988@185.28.212.118' AND `event`='dialog' Where 185.28.212.118 is the primary IP. In reality, it should be doing an additional query for anything considered `myself`. i.e. select `callid`,`to_tag`,`from_tag` from `active_watchers` where `presentity_uri`='sip:441164969988@10.20.11.118' AND `event`='dialog' I've been scouring the docs for a way to add additional IPs and/or domains for the purpose of querying active_watchers but I can't see anything relevant. What's the process here? Thanks Phil

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kamailio xmpp gateway configuration issue
by kevin hifx 12 Feb '16

12 Feb '16

Dear friends, I am facing an issue, when trying to configure xmpp gateway between kamailio and ejabbred xmpp server. I have followed the steps mention in this web site: http://www.kamailio.org/docs/modules/4.1.x/modules/xmpp.html The kamailio service is running but in the log file it show that xmpp connection failed. "*Feb 5 11:32:33 sip-xmpp-server /usr/sbin/kamailio[646]: ERROR: xmpp [network.c:122]: net_connect(): connect() failed: Connection refused* " (Both kamailio and ejabbred installed on the same server in local network and have local DNS server) I have added following entries in the kamailio configuration file. I am only mentioning the xmpp section. " ####### Global Parameters ######### alias=sip-xmpp.kevin.org ####### Modules Section ######## #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # XMPP loadmodule "xmpp.so" # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") modparam("rr", "enable_full_lr", 1) # XMPP modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "sip-xmpp.kevin.org") modparam("xmpp", "xmpp_domain", "xmpp-sip.kevin.org") modparam("xmpp", "xmpp_host", "xmpp.kevin.org") ###############XMPP # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; ### absorb retransmissions ### if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; ############### " could you please help me solve this issue. Thank you kevin

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Only Cassandra, Possible?
by Barış Şekerciler 12 Feb '16

12 Feb '16

Hello, I want to use only Cassandra DB for Kamailio. Is this possible? If it yes, how can I do that? For example, if I replace the DBENGINE as DB_CASSANDRA can I use it like this? Thank you. Regards.

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