sr-users December 2020

sr-users@lists.kamailio.org

67 participants
85 discussions

by James Browne

Has anyone ever implemented DMQ replication of TOPOS data between kamailio servers? I'm trying to find the easiest (to maintain) way to implement this replication, possibly without using an external database. James

2 years, 6 months

presence not work proper with subs_db_mode = 2

by Aatif Shaikh

Hello, all have a problem with Kamailio presence, I have configured my kamailio.cfg to handle dialog presence and facing issue in case of subs_db_mode = 2 in case of mode 2 facing the following presence issue. 1. after hangup presence is still up 2. sometimes stuck in a ringing state 3. sometimes presence not came..etc but in the case of subs_db_mode = 3 (DB ONLY) mode presence working fine not facing any issue with this mode. I noticed that sometimes Kamailio will throw the following message during PUBLISH processing and it will not generate appropriate NOTIFY I also verify inactive watcher subscription is there for that user and also a proper entry in the dialog table. MSG during PUBLISH processesing and it will not generate appropriate NOTIFY Oct 30 07:42:10 NC-HOSTED-1 /usr/local/sbin/kamailio[2409]: DEBUG: presence [notify.c:1444]: publ_notify(): Could not find subs_dialog kamailio.cfg param modparam("pua", "db_url", DBURL_KAMAILIO) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 100) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "default_expires", 3600) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", SERVER_IP) modparam("dialog", "db_url", DBURL_KAMAILIO ) modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 260) modparam("dialog", "db_fetch_rows", 500) modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", 8) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "use_pubruri_avps", 1) modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "override_lifetime", 300) modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") modparam("pua_dialoginfo", "pubruri_caller_dlg_var", "pubruri_caller") modparam("pua_dialoginfo", "pubruri_callee_dlg_var", "pubruri_callee") modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence", "db_table_lock_type", 0) modparam("presence", "db_url", DBURL_KAMAILIO) modparam("presence", "db_update_period", 5) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 100) modparam("presence", "subs_db_mode", 2) //modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 1) //new/ //modparam("presence", "notifier_processes", 2) //new// modparam("presence", "notifier_poll_rate", 20) //new// modparam("presence", "waitn_time",1) modparam("presence", "fetch_rows", 1000) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "max_expires", 3600) modparam("presence_xml", "db_url", DBURL_KAMAILIO) modparam("presence_xml", "force_active", 1) logic for presence request_route { if(is_method("PUBLISH") || is_method("SUBSCRIBE")) { route(HANDLE_PRESENCE); } } route[HANDLE_PRESENCE] { if (! t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } route[RELAY] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(t_ext) = $tU ; $avp(domain) = $fd ; $avp(et) = $_s(sip:$avp(t_ext)@$avp(domain)) ; $avp(s:puburis_callee) = $avp(et); setflag(8); dlg_manage(); } } I already report this issue in Kamailio git https://github.com/kamailio/kamailio/issues/2540 -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob/> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. 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2 years, 6 months

Dialog profile not defined

by Gertjan Wolzak

Hello Kamailions, I am running into the following issue. The same setup I had working on 5.0, now I am using a 5.4 version of Kamailio, but I do not believe it has to do with the version, more with my ability to make errors... I am trying to limit incoming calls by using the dialog module. The max concurrent calls value I retrieve from a database. But when I try to start Kamailio it will not because the dialog profile is not defined... These are my configured Dialog parameters: # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "callee") My check on the concurrent inbound calls: route[CONCURRENT_IN] { #Add call to callee profile #Check if callee has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) = $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("callee", "$avp(custid)"); return; } And the kamailio log error: Dec 3 09:12:08 proxy01 kamailio[74102]: INFO: <core> [main.c:2833]: main(): processes (at least): 21 - shm size: 67108864 - pkg size: 8388608 Dec 3 09:12:08 proxy01 kamailio[74102]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 3 09:12:08 proxy01 kamailio[74102]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Is someone able to see where I go wrong and point me in the right direction? Thank you. Rgds, Gertjan Wolzak

2 years, 6 months

$TF pseudo variable can not be used

by Володимир Іванець

Hello! I'm using a $TF variable in my config for some logging purposes. It was providing the correct string with Kamailio version 5.3.8 on CentOS 6 and 8. After updating to Kamailio version 5.4.1, the variable started to show something else. I also tried versions 5.4.2 and development 5.5.0 on CentOS 6 and the problem still persists. With version 5.5.0_dev3 this variable shows "pG▒" for me. I was wondering if someone else is using it too with Kamailio version 5.4.1 or newer on different platforms and has this issue. Thanks!

2 years, 6 months

kamailio51 installs 5.2.1?

by David Villasmil

Hello everyone, On a debian buster i added the repo deb http://deb.kamailio.org/kamailio51 buster main deb-src http://deb.kamailio.org/kamailio51 buster main But when i install i get 5.2, is this right? additional info: # apt-cache showpkg kamailio Package: kamailio Versions: 5.2.1-1 (/var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages) (/var/lib/dpkg/status) Description Language: File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages MD5: df3e15f422439e08c305782f5650a98c Description Language: en File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_i18n_Translation-en MD5: df3e15f422439e08c305782f5650a98c 5.1.10.1+bpo10 (/var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages) Description Language: File: /var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages MD5: ebddf40d0dfbfde1c479419970f978c7 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 years, 6 months

kamailio not absorbing ACK to CANCEL when behind a Path proxy

by Anthony Alba

Folks, I have a weird situation, when I enable Path header, kamailio behind a Path proxy, that kamailio doesn't recognise the ACK to CANCEL and tries to forward it. Test scenario: - voice.example.com is a stateless proxy with TLS/UDP bridging to kamailio. - on the UDP leg to kamailio it will add a Path: header to REGISTER UA1 = david is calling UA2 = charles and UA2 is sending 603 1 . UA1: preloaded route set (outbound proxy) # this is the Path proxy Route: sip:voice.example.com;transport=tls;lr 2. UA2: sends CANCEL 603 - kamailio immediately sends its (per-hop?) ACK - kamailio forwards CANCEL to UA1 kamailio sent this (per-hop?) ACK - the Route header is the internal UDP interface of the Path proxy ACK sip:charles@192.168.1.7:37309;transport=TCP;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.122.99:5064;branch=z9hG4bK735d.6b27c8d09e40462ef47eccf90eb10823.0 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: <sip:charles@voice.example.com>;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:192.168.122.100;lr> Content-Length: 0 3. UA1 sends ACK kamailio sees this - top Via is path proxy - 2nd Via is UA (caller) - CSeq correctly matches the INVITE - kamailio does not recognise this ACK from UA1(caller), tries to forward it, and also resends 603 to UA1(caller) 3 times ACK sip:charles@voice.example.com SIP/2.0 Via: SIP/2.0/UDP 192.168.122.100:5060;branch=z9hG4bKb3c9a7a3b6ec9e78d0144cda709f7047 Via: SIP/2.0/TLS 192.168.1.17:44925;rport=44925;branch=z9hG4bKPjFM-9gpaJEghJvN19TjnvScCwCoAGzwEc;alias;received=192.168.1.17 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: sip:charles@voice.example.com;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:voice.example.com;transport=tls;lr> Content-Length: 0 Now kamailio should absorb this ACK, as it has already sent its per-hop ACK. Instead I see ERROR: <core> [core/forward.c:541]: forward_request(): cannot forward to af 2, proto 3 no corresponding listening socket I don't expect kamailio to forward anything at this point. The route block is route[WITHINDLG] and it calls t_relay() normally. route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); # we should absorb this ACK, no? exit; } 4. If I remove the Path proxy and kamailio is the TLS proxy at voice.example.com the call flow works perfectly. The ACK from caller (david) is absorbed. 5. Regular calls, when UA2 accepts the call, work fine. In-dialog requests for BYE, from both UAs, work correctly with this Path proxy. Each UA has the complete 5 element route set( 2 x Path proxy + kamailio + 2 x Path proxy). The working route set looks like this: BYE sip:david@192.168.1.17:44925;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TLS 192.168.1.7:37309;rport;branch=z9hG4bKPjHdUtaUL5AfRcV-SX1Puya4niJKOOQ4VQ;alias Max-Forwards: 70 From: <sip:charles@voice.example.com>;tag=6zOt.H7w8NL5njV7klwGLlSJy.U5yBC9 To: <sip:david@voice.example.com>;tag=Pa0ORUxA-7lK0BnR4pLJ5n8rn2OvQrJy Call-ID: 3gWy4Q7O.-zQ-I07lRKHyFyX6azyCSwG CSeq: 12855 BYE Route: <sip:voice.example.com;transport=tls;lr;r2=on> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:192.168.122.99:5064;lr> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:voice.example.com;transport=tls;lr;r2=on> User-Agent: basesip 1.0.0 Content-Length: 0 Any ideas? Anthony Alba

2 years, 6 months

android client SIP TLS client autentication + kamailio

by beer Ll

Hello Has anybody use a android SIP client with TLS client authentication enabled with kamailio? Do you have suggestion ? Many thanks Luca

2 years, 6 months

Kamailio 5.4 ndb_redis, selecting a database

by Daniel Donoghue

Hello, I note that I can use the ndb_redis server config parameter to specify connection details for my redis servers; I also note that the connection string can specify a “db” parameter which can be set from 0 to 9, with the default being 0. I am looking for a way to dynamically set the db parameter. I do not which to add the same server with different names N times just to select a different db and having N times more connections to it than necessary. It looks as though I can call redis_cmd(“svrN”, “SELECT dbN”, “init”); but will that propagate properly and set the correct db for all connections to my redis server for the lifetime of the request? Ideally, I’d like to be able to specify an avp for the db, or have a function to set the db per request. With every blessing, — Daniel Donoghue

2 years, 6 months

Kamailio Remove Remove "Allow" features from header

by me.projects＠yahoo.com

Hi, I want to remove some "Allow" features from my Kamailio SBC like I want to keep following only Allow: OPTIONS, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, How can I achieve that? Thanks,

2 years, 6 months

How to DIY/Setup An Open Source IP PBX Appliance/Server?

by Turritopsis Dohrnii Teo En Ming

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celero… Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -----BEGIN EMAIL SIGNATURE----- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html ******************************************************************************************** Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -----END EMAIL SIGNATURE-----

2 years, 6 months

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sr-users December 2020