- sr-users - kamailio.org

issues with RTP on different hosts & dispatcher mod
by Infinicalls Infinicalls 02 Nov '15

02 Nov '15

Hi, I whenever I try to add rtpproxy to my config file modparam("nathelper", "rtpproxy_sock", "udp:firsthost:12221 udp:secondhost:12221") , I get this error. 0(45810) ERROR: <core> [modparam.c:139]: set_mod_param_regex(): parameter <rtpproxy_sock> of type <1> not found in module <nathelper> 0(45810) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 423, column 82: Can't set module parameter ERROR: bad config file (1 errors) 0(45810) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized I am having with_nat. Secondly, I tried with dispatcher and it works for one way calls. ie. User1 is able to call User2, but vice versa is not possible. Both RTP1 and RTP2 are in different machines and Kamailio sits on a different machine. All have identical network setups. And no NAT. Any idea. Can somebody guide me? thanks. regards Ganesh Kumar

2 1

100% CPU usage after NTP date update
by Stefan Kohlhauser 02 Nov '15

02 Nov '15

Hello everyone! Setup: I am currently using a Kamailio 4.2.3 on an embedded device (low CPU and RAM). The device is not able to store the current date, so after a reboot its 1. Jan 1970 until NTP updates the date. Problem: If the date is updated after the Kamailio has started (a leap of 45 years) the Kamailio uses up all the CPU and renders the device useless for a very long time. Notes: There are no incomming requests at the time. (But the Kamailio is not responsive during that time anyway.) Adjusting the internal Kamailio-time seems to take longer when the date update leap is bigger. My questions: 1) Can I prevent the Kamailio from using up the entire CPU after a date update? Or maybe do the internal ticks-update more gracefully? 2) For my better understanding: What happens within the Kamailio after adjust_ticks has updated the internal date? Thanks in advance for your help! Best regards, Stefan

2 4

Multi-Array with AVPs
by Jonathan Hunter 02 Nov '15

02 Nov '15

Hi Guys, Hope all are well? sorry to post but Im just trying to get something working where Im trying to convert a php function script into the kamailio.cfg. Im getting there but wonder how I can group results of sets of avps with another avp. Say I have these variables; $(avp(NTgrp)) = 'SB100_000361@1.2.3.4|SB100_000213@1.2.3.5|SB100_000225@1.2.3.6|SB100_000361@1.2.3.7|SB100_000213@1.2.3.8';$(avp(Prio)) = '1|1|1|2|2';$(avp(Wgt)) = '10000|10000|80000|30000|70000';$(avp(CInc)) = '0|0|1|0|1';$(avp(WTot))= '100000|100000|100000|100000|100000'; I then run a while script to split up their variables via the delimiter | in user; $var(j)=0;while($var(j)<$var(count)) {$(avp(NTgrpP))=$(avp(NTgrp){s.select,$var(j),|});$(avp(PrioP))=$(avp(Prio){s.select,$var(j),|});$(avp(WgtP))=$(avp(Wgt){s.select,$var(j),|});$(avp(CIncP))=$(avp(CInc){s.select,$var(j),|});$(avp(WTotP))=$(avp(WTot){s.select,$var(j),|});xlog("L_INFO"," Now Each time we have $(avp(NTgrpP)) $(avp(PrioP)) $(avp(WgtP)) $(avp(CIncP)) $(avp(WTotP))\n");$var(j)=$var(j)+1;} This then results in;INFO: <script>: Now Each time we have SB100_000223(a)1.2.3.4 1 10000 0 100000INFO: <script>: Now Each time we have SB100_000225(a)1.2.3.5 1 10000 0 100000INFO: <script>: Now Each time we have SB100_000386(a)1.2.3.6 1 80000 0 100000INFO: <script>: Now Each time we have SB100_000213(a)1.2.3.7 2 30000 0 100000INFO: <script>: Now Each time we have SB100_000361(a)1.2.3.8 2 70000 0 100000 Now what Id like to do is group them via priorities so I can do some further functional checks on them when in same priority group, as you can see there are 3 results at priority 1 and 2 at priority 2. Is there away I can create something of a multi-array using the AVP $(avp(PrioP)) I just created so I can group by priority? As in php I would do something like; $multi_arr[$prio_arr[$p]][] = array( "ntgrp" => $ntgrp_arr[$p], "prio" => $prio_arr[$p], "wght" => $wght_arr[$p], "cinc" => $cinc_arr[$p], "wtot" => $wtot_arr[$p], ); } Any help would be great. Many thanks Jon

3 6

Q: about CRL list (TLS)
by Vladimer Gabunia 02 Nov '15

02 Nov '15

hello all. we compiled kamailio with TLS Support. but have next problem when using CRL Lits. Our Certificate issuing scheme is follow: Offline Root CA -> Enterprise SubCA -> Server and Phone Certificate CRL list is signed by SubCA. option "require client certificate is enables (1) " When we enable CRL list, phones are not registered. CA file is offline RootCA certificate in pem format. We think that the reason is that СRL was signed by Subca or incorrect CRL format. CRL is converted from MS CRL to PEM. (What is the format for the CRL) maybe someone have experiance with similar scenarios? thanks in advance!! ________________________________ [gh.ge] ვლადიმერ გაბუნია IT სამსახურის უფროსი ტელ: (+995) 32 2505222 +8183 მობ: (995) 577 095333 შპს "ჯეო ჰოსპიტალს" სათავო ოფისი თბილისი 0160, ვაჟა-ფშაველას გამზ. № 16; http://www.gh.ge <http://gh.ge>

2 5

Problems with dispatcher module | feature disabled
by Alessio Casco 02 Nov '15

02 Nov '15

Hello! We noticed today an issue with the kamailio loadbalancing. Basically kamailio is not considering if an host is up or not, backends are set always as "FLAGS: AP" even if the host is down. I see this when the proxy starts : ERROR: dispatcher [dispatcher.c:768]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled These the parameters I have set: # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;code=404") modparam("dispatcher", "ds_ping_from", "sip:proxy@sip.nexmo.com") modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_hash_size", 10) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 10) modparam("dispatcher", "use_default", 0) modparam("dispatcher", "attrs_pvname", "$avp(carrierattr)") modparam("dispatcher", "setid_pvname", "$avp(carriergroup)") modparam("dispatcher", "flags", 2) I use 4.3.3, I think this issue started after upgrading from 4.3.1 but I can't say for sure. Can someone please give us some advices on this? Thanks Alessio

2 5

Which option uses for check certificate revocation in kamailio 1.5.x ?
by Evgeniy Ivanov 02 Nov '15

02 Nov '15

Hello! I use kamailio 1.5.4-tls (i386/linux). I configured the tls options: disable_tls = 0 listen=tls:192.168.50.60:5061 tls_certificate="/etc/kamailio/tls/crt.pem" tls_private_key="/etc/kamailio/tls/key.pem" tls_ca_list="/etc/kamailio/tls/ca.pem" tls_verify_client=on tls_require_client_certificate=on I need to check certificate revocation, but the option tls_crl does not work. Which option uses for check certificate revocation in kamailio 1.5.x ?

3 6

Display name
by Loïc Boissy 30 Oct '15

30 Oct '15

Hello, I would like to know if kamailio is able to store contact "display name" in its database? Thank you for your help. Best regards. Loïc

2 2

Routing outgoing call from asterisk to registered SIP provider
by Bruno Salzano 29 Oct '15

29 Oct '15

Hello, i've DID provider that require to register to it's server. I've been able to configure this provider using kamailio and uac and to route all inbound calls from DID provider to internal asterisk media server. All is working great. Now i need help to configure call routing from internal asterisk to the external DID provider. I've done something like: route[LOCATION] { if(is_method("INVITE")) { route(FROMASTERISK); route(TOASTERISK); exit; } [...] } route[FROMASTERISK] { if (ds_is_from_list()) { xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server Cluster\n"); rtpengine_manage("direction=internal direction=external replace-origin replace-session-connection trust-address"); $ru = "sip:TESTING_NUMBER@EXTERNAL_PROVIDER_IP"; route(RELAY); exit; } } It seems that when i route the call outbound, i get a 407. So i would like to react to this using uac_auth() as described here: http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth() Is this the right way? Can someone help me, giving hints and some working example? Using kamailio 4.3.3 with rtpengine 4.1.0 Best regards, Bruno -- The life is short.. live at your best!

4 6

gcc warning for t_cancel
by Yasin CANER 27 Oct '15

27 Oct '15

3 3

Routing between 2 domains
by kai.ohnacker＠cbc.de 27 Oct '15

27 Oct '15

Hello community, I work with Kamailio 4.2.0 which is placed between two networks (with two interfaces) and RTPproxy in bridge mode. The task from Kamailio is to handle the calls from internal networks to internal networks only on the internal Interface without bridging. For external networks vice versa. If the call comes from external to internal, then the RTPproxy should work in bridge mode and vice versa. I adapt the example with IPv4 and IPv6 routing, but this is not working for internal to external calls and vice versa (http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). Only external calls and only internal calls works. I think the cause of this behavior is the routing which I implemented to the RTPmanage part in the Kamailio config. The error message is given by the internal device. (call from external to internal) If I make a call from internal to external, the RTP stream is only in one direction. Internal Interface IP: 203.207.111.58 External Interface IP: 193.16.163.58 The RTPproxy config is: CONTROL_SOCK=udp:127.0.0.1:9000 EXTRA_OPTS="-l 203.207.111.58/193.16.163.58 -d WARN:LOG_LOCAL1" Kamailio.cfg: #!define INT_IP 10.96.0.0/14 #!define EXT_IP 193.0.0.0/8 # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; # rtpproxy_manage("co"); # Start Test routing if((src_ip==INT_IP && dst_ip==EXT_IP)){ rtpproxy_manage("cowie"); } if((src_ip==EXT_IP && dst_ip==INT_IP)){ rtpproxy_manage("cowei"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } Can someone help me with the right routing? Do you need some more debugging information? Cheers, Kai

2 4

sqlite errors
by nadie 27 Oct '15

27 Oct '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, sorry for the question. Kamailio 4.3.3, debian Jessie: Kamailio with sqlite all work correctly but syslog tells: Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [ucontact.c:683]: db_insert_ucontact(): inserting contact in db failed Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [urecord.c:418]: wb_timer(): inserting contact into database failed (aor: 17) Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:489]: db_sqlite_commit(): sqlite commit failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:138]: db_sqlite_cleanup_query(): finalize failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [ucontact.c:683]: db_insert_ucontact(): inserting contact in db failed Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [urecord.c:418]: wb_timer(): inserting contact into database failed (aor: 50) Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:489]: db_sqlite_commit(): sqlite commit failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:138]: db_sqlite_cleanup_query(): finalize failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [ucontact.c:683]: db_insert_ucontact(): inserting contact in db failed Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [urecord.c:418]: wb_timer(): inserting contact into database failed (aor: 59) kamailio.cfg #!KAMAILIO #!define WITH_SQLITE #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS ..... #!ifdef WITH_SQLITE loadmodule "db_sqlite.so" #!endif ...... #!ifdef WITH_SQLITE modparam("auth_db", "db_url", "sqlite:///home/kami/app/kamailio.sqlite") #!define DBURL "sqlite:///home/kami/app/kamailio.sqlite" #!endif ............ # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJWLnRoAAoJEG5DH1ilMeee+P8P/A3/YiDqiA2VsZR3GvTtAYgh 5XjGBI9MptT2IuLPe3FMnuOtKPGBkKIlcUpW8EmAvV37wVooHl0ZyF4/Ap+izgMm G8PeoOhgQ0jaY+j+p/KVK3OpN3uQprH/F+H0o5ljHi2R9oYE/FFVv8/A+mGcAztD KEmqSq9HNqcCn+4p+qB5yMM+nmi9vCGWFlPiMIInQ7YOIWLCkJJCjkeGgjKgQzJw tAYRJkoytmpTbUxTw7bwPYniQee7ATfXaMhslLU0bIGIleFjuElcC32NoBs6KZKW jYAWjMMwuVtyMxjkLq29q46/Me/D7PIbIBuxxoPUho9OKCjEN3ob2K3GRvJf6d5j p0uNYCNY6nsftdO3UzZE1+xz9NqXvpmO+oOcWw/J9N+5/vMEOorwwvmoD5JX4LSn 7jqXkeGvGoqiwPkquLEYfTWB0hOZJyDRHJWej9WBTPF063seLvaX5OFRXjuvOIHd D6fmG4vxMTB0VpANxQQuf73HA0hlYZ5WzvjoNT8cc259uWY8OPiuLcIuOfHtBphk xlttHbTnozCvCJVP4W0OLDZHwPR3AuOdBhtjOM+kIsUwzMVlVA5/Drqai/EDsX4/ tzVPHKNneBU98jPtNTmFQqWnT2sGnZeNYBO/HdcZYNGFsJ993WCQvVAZZ/0OwL+J hV3EXhaP/KqeZRtHjwFN =x7Zq -----END PGP SIGNATURE-----

2 1

Kamailio start up fails with strange IO Error on RHEL 7.1
by M S 26 Oct '15

26 Oct '15

Hi, I have been trying to setup Kamailio v4.3.3 rev. e275bc on RHEL 7.1 64bit. It has been very difficult to resolve all dependencies initially due to RHEL's extremely limited repositories and libraries support, but nevertheless I am finally able to successfully compile kamailio with all essential modules on it. However, now kamailio fails to start up and gives this strange error, -- 36(3335) CRITICAL: <core> [io_wait.c:526]: init_io_wait(): could not alloc epoll array 36(3335) CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... 0(3250) ALERT: <core> [main.c:725]: handle_sigs(): child process 3335 exited normally, status=255 0(3250) INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD -- I have never seem this error before nor i can find any info related to it anywhere on internet. So, i doubt it is due to any missing dependency. Any idea what may be the cause of this? Thank you.

2 6

Unable to Update Headers in failure route which were build in request route
by Sunil More 26 Oct '15

26 Oct '15

Hi All, I am trying to have a failover route where if a initial relay fails , i can forward the same request to a second provider. I am using a certain set of headers that help to authenticate over my providers. I am trying to update those headers. However update and removal both fail. The headers are still present. Please help me understand if i am wrong. Here's the code. ..... route(GET_CREDENTIALS); } remove_hf_re("X-*"); append_hf("X-Src-IP: Trusted\r\n", "Call-ID"); append_hf("X-Identity: $hdr(X-Calling-User)\r\n", "Call-ID"); append_hf("X-Auth-ID: $avp(siptrunk_username)\r\n", "Call-ID"); $avp(realm) = $fd; $rd = $avp(siptrunk_domain); $var(duri) = "sip:"+$rU+"@"+$avp(siptrunk_proxy); $du = $var(duri); t_on_failure("UAC_AUTH"); if(!t_relay()) { sl_reply_error(); exit; } } failure_route[UAC_AUTH] { if(t_check_status("(4|5)[0|8][0|2|3|4|6|8]")){ xlog("L_INFO","failover testing"); revert_uri(); route(FAILOVER); } if (t_check_status("40[17]")) { xlog("got challenged \n"); uac_auth(); xlog("L_INFO", "Return code is $retcode"); $rd = $avp(siptrunk_proxy); t_on_failure("UAC_AUTH"); t_relay(); } } route[FAILOVER]{ xlog("L_INFO","doing failover now.. "); if($avp(failover_routing_id)!=$null) { $avp(outbound_routing_id)=$avp(failover_routing_id); xlog("L_INFO","Failover carrier credentials are.."); route(GET_CREDENTIALS); #not adding headers now remove_hf_re("X-*"); append_hf("X-Src-IP: Trusted\r\n", "Call-ID"); append_hf("X-Identity: $hdr(X-Calling-User)\r\n", "Call-ID"); append_hf("X-Auth-ID: $avp(siptrunk_username)\r\n", "Call-ID"); $avp(realm) = $fd; $rd = $avp(siptrunk_proxy); $var(duri) = "sip:"+$rU+"@"+$avp(siptrunk_proxy); $du = $var(duri); xlog("L_INFO","Destination Uri from failover is : $du"); t_on_failure("UAC_AUTH2"); if(!t_relay()) { sl_reply_error(); exit; } exit; } } failure_route[UAC_AUTH2] { if(t_check_status("(4|5)[0|8][0|2|3|4|6|8]")){ xlog("L_INFO","Failover failed. cancelled call "); exit; } if (t_check_status("40[17]")) { xlog("got challenged \n"); uac_auth(); xlog("L_INFO", "Return code is $retcode"); $rd = $avp(siptrunk_proxy); t_relay(); } exit; } Thanking You, Sunil More

3 2

issue with sending "delete" to rtpengine for cancelled branch
by Vasiliy Ganchev 26 Oct '15

26 Oct '15

Hi folk! Have an issue, and do not see the obvious way how to solve it, maybe i'm missing something. The call scenario: UA-A (hard phone, no rtpengine) call ->> UA-B (has 2 registration: hard phone (no rtpengine) and WebRTC based web phone (here the rtpengine is called, in branch_route)) When Kamailio fork call to UA-B, for the branch with WebRTC UA, I do rtpengine_manage (rtpengine receive command "offer" and allocate 4 ports). If UA-B answer via hard phone, Kamailio Cancel branch to WebRTC phone, and here I need to do rtpengine_manage (to send "delete" command to rtpengine). But as I see Kamailio CANCEL all nonanswered branches automatically (I see no place where to call rtpengine_delete) Any suggestion how to cope with such issue? cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/issue-with-sending-delete-to-rtpeng… Sent from the Users mailing list archive at Nabble.com.

1 1

kamctl rtpproxy nh_show_rtpp not working
by amar Smart Telecom 25 Oct '15

25 Oct '15

Dear all, I have installed Kamailio and Rtpproxy. Following is the process (ps) info: "kamailio 648 1 0 Oct08 ? 00:00:05 /usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.58.25" But when I give command: " kamctl mi nh_show_rtpp" it returns following: "500 command 'nh_show_rtpp' not available" What could be the problem? Thanks, Amar.

3 5

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://vivamous.com/youth.php?lkb> brettlist(a)nemeroff.com

2 1

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://stillbontechnology.com/smiling.php?h98ha> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://janatyachar.org/woman.php?mq> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://moverdubai.net/having.php?tky> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://lapeste.org/land.php?eiz> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://igoasia.org/wish.php?9w> brettlist(a)nemeroff.com

1 0

msilo dump with non 2xx answer
by Péter Barabás 22 Oct '15

22 Oct '15

Hi, I set instant message store&forward in kamailio using MSILO module. It works fine in many cases. We have special message handing (encryption in "session"), i do not go into detail since it is not necessarily to solve the problem. So there are situations, when the offline receiver should get a MESSAGE and will acknowledge with 4xx status code, like 493: the original message is stored in silo table and when the receiver will be online, it will be dumped and he will send 493. Question: how can I delete the message from silo table with MSILO (since it is not 2xx) and how can i forward this ack message to the original sender instead kamailio? Thanks. Dr. Péter Barabás Director of Software Development Arenim Technologies AB.

3 4

Kamailio and FreeSwitch - ClueCon Weekly Conference Call
by Daniel-Constantin Mierla 21 Oct '15

21 Oct '15

On Wednesday, October 21, 2015, at 17:00GMT (12:00CT, 18:00 London, 19:00 Berlin), the Cluecon weekly conference call will focus on Kamailio and FreeSwitch. I will be joining the call, answering the questions about Kamailio and its options to integrate with FreeSwitch. Expect the FreeSwitch core developers to be around to handle the questions about their project. Participation is open for anyone, you can dial in for audio or video sessions using a SIP phone or webrtc capable browser: - sip:888@conference.freeswitch.org - https://cantina.freeswitch.org/vc More dial in options (e.g., PSTN) are presented at: - https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

MSILO module
by Safdar Khan 21 Oct '15

21 Oct '15

How can i store messages in silo table not only offline also online.Demo code will be appreciated.waiting for response.

2 3

htable never
by Juha Heinanen 21 Oct '15

21 Oct '15

i noticed that entries that i had added to htable at sip proxy start in event_route [htable:mod-init], e.g., $sht(htable=>app_srv::sbc::<null>) = "sip:SBC:SEMS_PORT"; had disappeared from htable after some (don't know exactly how long) time. i have not specified any autoexpire for that table: modparam("htable", "htable", "htable=>size=HTABLE_SIZE;dbtable=htable;") after restart of sip proxy they reappeared in htable like this: # sip-proxy_ctl htable.get htable 'app_srv::sbc::<null>' { item: { name: app_srv::sbc::<null> value: sip:127.0.0.1:5090 flags: 2 expire: NEVER } } what could explain the disappearance? -- juha

3 3

Implementation of RFC 5393
by Guillaume 21 Oct '15

21 Oct '15

Hi guys, What do you think about the RFC 5393 on loop detection and amplification attack protection? The RFC is short and still a proposed standard but don't you think it could be useful to prevent loop and amplification attack? Because even if the max-forward field reduces the loop to ~70 hosts (in most cases) with some techniques we could fork the message up to 2^70 messages (as described in the RFC) to crash the servers. Basically the server has to do 2 things: * check if it is not already in the via of the message * the previous check is not enough as a B2BUA could have replace the via headers, so the RFC introduces a new field called max-breadth to limit the forking. I have not seen a lot of implementation of this RFC on the free SIP software and I think it could be a good way to improve kamailio making a module for it (the easier way to implement this feature I think). In fact I'm in a research internship about VoIP security and I have time to develop such a module for kamailio if you think it's a good idea (I'm looking for some security improvements in free software solutions so if you have other idea don't hesitate to tell me). Cheers, Tetram

3 4

pua_dialoginfo publish sent without recieved etag (intermittent)
by Asgaroth 21 Oct '15

21 Oct '15

Hi All, I've come accross an intermittent issue where an initial publish is sent to our presence server, proxy recieves the subsequent 200 with etag, but the following publish sent does not contain the sip-if-match header of the recieved 200, which ends up causing trouble with BLF on our test devices, it also has the added side affect of not being cleared out of the presentity table, these entries need to be manually removed. This does not occur all the time, I can use the same phone and make several calls before this issue occurs. It is not limited to a device type, we've tried with panasonic, polycom, cisco, zoiper, yealink and linksys devices, all exhibit the same sporadic behaviour. I've attached an example sip trace of the publish messages where this happens, you can see the second publish doesnt have the sip-if-match header whose content should be that of the sip-etag in the previous 200 to the initial publish. I'm not sure if there is something I am misunderstanding here, a configuration issue, or if this is indeed a bug. Both the proxy and the presence server are running kamailio v4.3.3. Any pointers/suggestions/guidance would be appreciated. If you need any additional info, please dont hesitate to ask. Thanks

2 2

Sending ReINVITE from Kamailio
by Grant Bagdasarian 21 Oct '15

21 Oct '15

Hello, Is it possible to have Kamailio send a ReINVITE every X minutes to determine if a session is still active? I know it's a proxy and doesn't have B2BUA capabilities, but there was a module which allowed Kamailio to generate SIP messages, but I can't find it anymore. If Kamailio is not the place to do this, which component in the voip network should be responsible for this? Are there perhaps other ways to poll for session state in Kamailio? Regards, Grant

3 4

Re: [SR-Users] MySQL Connect Errors
by John Billings 20 Oct '15

20 Oct '15

Both boxes are running Debian 8. I’m still looking at the mysql logs, but I haven’t found anything relevant, yet. > On Oct 14, 2015, at 5:00 AM, sr-users-request(a)lists.sip-router.org wrote: > > Hello, > > do you have centos+selinux? Because I heard about a lot of limits with > the combination of the two. > > Have you checked the mysqld syslog messages, anything relevant there? > > Cheers, > Daniel >

3 2

Far end NAT traversal
by NITESH BANSAL 20 Oct '15

20 Oct '15

Hello, I would really appreciate if someone can point me to some documentation for Far end nat traversal in Kamailio. Thanks,Nitesh

3 2

Dialog timeout if record route have both strict and loose route
by Arun Kumar 20 Oct '15

20 Oct '15

Hi dialog timeout is occurred if 200 ok having mixed loose and strict routing entries in record route header conf file attached INVITE sip:123456789@22.22.22.22:5060 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjvAKLQD72G88kpqC2IJwbmTcXUlHGULcW Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22 Contact: <sip:123455555@176.204.26.157:63939;ob> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14602 INVITE Route: <sip:11.11.11.11:345;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: zeo-iphone Opcode: 19815 Content-Type: application/sdp Content-Length: 320 v=0 o=- 3653999880 3653999880 IN IP4 10.220.83.191 s=pjmedia b=AS:25 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 18 96 c=IN IP4 10.220.83.191 b=TIAS:8000 a=rtcp:4001 IN IP4 10.123.11.223 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=X-adapter:some value SIP/2.0 407 Proxy Authentication required Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK3d4f.6f8dff1ec0806cb23f56d847d5fe3934.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjvAKLQD72G88kpqC2IJwbmTcXUlHGULcW;received=176.204.26.157 From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: <sip:123456789@22.22.22.22> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14602 INVITE Proxy-Authenticate: Digest nonce="6b24dc6996de20028c78dbeaf13edc55",realm="22.22.22.22",opaque="",stale=FALSE,algorithm=MD5 Content-Length: 0 ACK sip:123456789@22.22.22.22:5060 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjvAKLQD72G88kpqC2IJwbmTcXUlHGULcW Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22 Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14602 ACK Route: <sip:11.11.11.11:345;lr> Content-Length: 0 INVITE sip:123456789@22.22.22.22:5060 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22 Contact: <sip:123455555@176.204.26.157:63939;ob> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Route: <sip:11.11.11.11:345;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: zeo-iphone Opcode: 19815 Proxy-Authorization: Digest username="123455555", realm="22.22.22.22", nonce="6b24dc6996de20028c78dbeaf13edc55", uri=" sip:123456789@22.22.22.22:5060", response="554a1da255084f7a95d3b6e7a0c48933", algorithm=MD5 Content-Type: application/sdp Content-Length: 320 v=0 o=- 3653999880 3653999880 IN IP4 10.220.83.191 s=pjmedia b=AS:25 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 18 96 c=IN IP4 10.220.83.191 b=TIAS:8000 a=rtcp:4001 IN IP4 34.220.83.191 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=X-adapter:some value SIP/2.0 100 Trying Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: <sip:123456789@22.22.22.22> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Content-Length: 0 SIP/2.0 183 Session progress Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: <sip:123456789@22.22.22.22>;tag=GB-1445011078679 CSeq: 14603 INVITE Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh Contact: <sip:22.22.22.22:5060> Content-Type: application/sdp Record-Route: <sip:22.22.22.22:5060;lr>,<sip:11.11.11.11:345;lr> Content-Length: 298 v=0 o=- 3653999880 3653999880 IN IP4 50.56.154.204 s=zeodialer c=IN IP4 50.33.154.204 b=AS:25 t=0 0 m=audio 53128 RTP/AVP 18 96 c=IN IP4 50.33.154.204 b=TIAS:8000 a=rtpmap:18 G729/8000 a=sendrecv a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=X-adapter:some value SIP/2.0 183 Session Progress Record-Route: <sip:22.22.22.22:5060;lr>,<sip:11.11.11.11:345> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE P-Early-Media: sendrecv Contact: <sip:22.22.22.22:5060> Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 To: <sip:123456789@22.22.22.22>;tag=GB-1445011078679 Content-Type: application/sdp From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu Content-Length: 212 v=0 o=- 1426744035 1426744035 IN IP4 50.56.154.204 s=Nexge c=IN IP4 50.22.154.204 t=0 0 m=audio 53128 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=ptime:20 SIP/2.0 200 OK Record-Route: <sip:22.22.22.22:5060;lr>,<sip:11.11.11.11:345> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE Contact: <sip:22.22.22.22:5060> Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 To: <sip:123456789@22.22.22.22>;tag=GB-1445011078679 Content-Type: application/sdp From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu Content-Length: 212 v=0 o=- 1426744035 1426744035 IN IP4 50.56.154.204 s=Nexge c=IN IP4 50.33.44.204 t=0 0 m=audio 53128 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=ptime:20 ACK sip:11.11.11.11:345 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjldbprXw3ybGOxoeKWwvtdxLmNp8jXsOX Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22;tag=GB-1445011078679 Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 ACK Route: <sip:22.22.22.22:5060;lr> Route: <sip:22.22.22.22:5060> Content-Length: 0

2 3

kamctl rtpproxy nh_show_rtpp not working
by amar Smart Telecom 19 Oct '15

19 Oct '15

Dear all, I have installed Kamailio and Rtpproxy. Following is the process (ps) info: "kamailio 648 1 0 Oct08 ? 00:00:05 /usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.58.25" But when I give command: " kamctl mi nh_show_rtpp" it returns following: "500 command 'nh_show_rtpp' not available" What could be the problem? Thanks, Amar.

2 1

RtpProxy socket timeout not working
by Arun Kumar 19 Oct '15

19 Oct '15

Hi Rtp Session timeout notification is not working , we tried with kamailio 4.2 with rtpproxy 2.0 / 1.3 beta/1.2 releases , i think kamailio rtpproxy is not submitting session timeout socket to rtpproxy kamailio 4.2.0 ----------------------------------- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:5060/RPC") ............... rtpproxy_manage(); ................ RtpProxy ------------------------- RTPProxy 2.0 RTPProxy 1.3 Beta with patch given by *Daniel* RTPProxy 1.2 rtpproxy -F -s udp:127.0.0.1:7722 -d DBUG -f -i -T 10 RTPRoxy Log ---------------------------------------------------- DBUG:get_command: received command "4242_9 Uc0,8,101 2479e562440c023d 192.168.1.103 6074 5410ef73;1" INFO:handle_command: new session 2479e562440c023d, tag 5410ef73;1 requested, type strong INFO:handle_command: new session on a port 45984 created, tag 5410ef73;1 INFO:handle_command: pre-filling caller's address with 192.168.1.103:6074 DBUG:doreply: sending reply "4242_9 45984 " DBUG:get_command: received command "4242_10 Lc0,101 2479e562440c023d 192.168.1.102 4002 5410ef73;1 ADqH-A-mLLN0th2gGoopYMK1cy.-ozDf;1" INFO:handle_command: lookup on ports 45984/36756, session timer restarted INFO:handle_command: pre-filling callee's address with 192.168.1.102:4002 DBUG:doreply: sending reply "4242_10 36756 " INFO:rxmit_packets: callee's address latched in: 192.168.1.102:4002 (RTP) INFO:rxmit_packets: callee's address latched in: 192.168.1.102:4003 (RTCP) INFO:rxmit_packets: caller's address latched in: 192.168.1.103:6075 (RTCP) INFO:rxmit_packets: caller's address latched in: 192.168.1.103:6074 (RTP) INFO:process_rtp: session timeout INFO:remove_session: RTP stats: 328 in from callee, 5 in from caller, 333 relayed, 0 dropped INFO:remove_session: RTCP stats: 3 in from callee, 4 in from caller, 7 relayed, 0 dropped INFO:remove_session: session on ports 45984/36756 is cleaned up Kamailio RTP log ----------------------------------------------- 3(4242) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 3(4242) DEBUG: rtpproxy [rtpproxy.c:2704]: force_rtp_proxy(): proxy reply: 45984 for more log please see attachment

2 1

Kamctl cannot add user to pgsql
by Pavlo Karbovnyk 19 Oct '15

19 Oct '15

Good Day All. I run into following problem: (It may be helpful for other people how have similar problem) kamctl add 1000 password database engine 'PGSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 <--- showing counter=0 check_db_alias: alias counter= <--- showing counter= nothing INFO: user '1000' already exists as alias ============================================================ how to fix it : in kamctl locate a function check_db_alias() { ..... CNT=`$DBROCMD "$QUERY" | $EGREP -v ERROR | $LAST_LINE` ..... } replace with CNT command with following: check_db_alias() { ..... CNT=`$DBROCMD "$QUERY" "$DBRAWPARAMS" | $EGREP -v ERROR | $LAST_LINE` ..... } ============================================================ after this kamctl add 1000 password database engine 'PGSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 pgsql_query: /usr/bin/psql -A -q -t -P fieldsep=' ' -h localhost -U kamailio kamailio -c 'insert into subscriber (username, domain,ha1,ha1b,password) values ('1000','127.0.0.1','6fda0594de6ea94b03a12fdd998758ae','31e118ac5b09f07 1858740d6125d901d','');' new user '1000' added Regards Pavlo.

2 1

Weight balancing with dispatcher module
by Ben Bliss 19 Oct '15

19 Oct '15

Hi, I am having issues getting the dispatcher to work using weighting. I have been digging through old posts to the list, and tried all that I have found, but am still having no luck at all. So, information... version: kamailio 4.2.4 (amd64/freebsd) be62bd flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, select, kqueue. id: be62bd Dispatcher configuration... modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 3) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") Dialplan config.. route[DISPATCH] { if(!ds_select_dst("1", "9")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; I am loading the values from a database, which are being loaded fine, and appear to be being formatted correctly aswell.. +----+-------+-------------------------+-------+----------+------------+----------------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+-------------------------+-------+----------+------------+----------------------+ | 1 | 1 | sipx.x.x.106:5060 | 8 | 1 | weight=100 | | | 2 | 1 | sip:x.x.x.107:5060 | 8 | 1 | weight=0 | | +----+-------+-------------------------+-------+----------+------------+----------------------+ The IP's are public, so i've masked them... The following is from kamcmd.. { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:x.x.x.107:5060 FLAGS: AP PRIORITY: 1 ATTRS: { BODY: weight=0 DUID: MAXLOAD: 0 WEIGHT: 0 } } DEST: { URI: sip:x.x.x106:5060 FLAGS: AP PRIORITY: 1 ATTRS: { BODY: weight=100 DUID: MAXLOAD: 0 WEIGHT: 100 } } } } } } However, when any call is placed, it will always goto the server ending 107 (with weight 0, which means, as far as I understand, it should be ignored). I've tried with ; after the weights, this made no change either. When set to round robin, or via priority, it appears to work fine. Any ideas? -- Kind Regards, --------- Ben

2 3

kamailio cfg file with user db authentication and rtpproxy
by amar Smart Telecom 19 Oct '15

19 Oct '15

Dear all, Can you pls mail me a working Kamailio cfg file which has: "Sip user db authentication" and "rtpproxy for all calls" Thank you, Amar.

3 2

Link between presence module and subscriber table
by Loïc Boissy 19 Oct '15

19 Oct '15

Hello, l would like to send a SIP message (NOTIFY for example) to a specific user when a new account is created in the kamailio database. Is it possible to do that? Best regards Loïc

2 1

Re: [SR-Users] call-limit peer
by Angel Elena 19 Oct '15

19 Oct '15

Ok, now I check dlg_manage settings. Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ -------------------------------- -----Mensaje original----- De: ycaner <yasin.caner(a)netgsm.com.tr> Enviado: Lun 19-10-2015 08:22 Asunto: Re: [SR-Users] call-limit peer Para: sr-users(a)lists.sip-router.org; > Hello; > with dialog module using dont forget set dlg_manage or setting flags. > or you can do it with pike limit for only INVITE methods. > > > > > -- > View this message in context: > http://sip-router.1086192.n5.nabble.com/call-limit-peer-tp142403p142407.html > Sent from the Users mailing list archive at Nabble.com. > > _______________________________________________ > 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 >

1 0

call-limit peer
by Angel Elena 19 Oct '15

19 Oct '15

Hi everybody. I have a kamailio 4.0 running with rtpproxy and debian 7.8, and I need limit calls by IP. My kamailio.cfg: # ----------------------------------------- #------------------------------------------ loadmodule "dialog.so" modparam("dialog", "dlg_flag", 4) modparam("dialog", "enable_stats",1) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) In my route outside, I have: get_profile_size("callquota", "$avp(s:tag)", "$var(SIZE)"); xlog ("L_INFO", ".... $avp(s:tag) ..... CALLQUOTA .... $var(SIZE)..... \n "); On $avp(s:tag) is the client code-peer/ip. When I view the logs... the result: Oct 18 22:18:03 sipproxy /usr/local/sbin/kamailio[19724]: INFO: <script>: .... 4012 ..... CALLQUOTA .... 0..... Always, the result is '0'. Is correct my configuration ? thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ --------------------------------

2 1

Re: [SR-Users] Changing a user's password in the Kamailio routing logic
by Surendra.P 19 Oct '15

19 Oct '15

Hello mike, you can achieve it by doing DB update query using sqlops module. If you want to change the user's password dynamically under a certain condition, just invoke a update query to the DB using sqlops. Hope it solves your issue. Regards Surendra Hi all, I was wondering if there was a way to change a user's password from within the routing logic section of kamailio.cfg (i.e. like do what 'kamctl password' does, but in the routing logic)? Thank, Mike ******

1 0

Re: [SR-Users] Auth module Issue:UA getting challanged thrice when load
by Surendra.P 19 Oct '15

19 Oct '15

Hello Daniel, Thanks for your reply. I did some debugging, I found that I am getting error like this, earlier this log was in DBG mode, changed to error and then I ran the load.. "auth:pre_auth: Invalid nonce value received (ret 2)" Problem is that this issue getting simulated every time. I am using auth module version of kamailio 4.1.5 , since I checked with latest kamailio version 4.3.2 , I didn't find any differences in this module. Please find my finding below Auth/nonce.c:check_nonce() b_nonce2_len=calc_bin_nonce_md5(&b_nonce2, cfg, secret1, secret2, msg); if (!memcmp(&b_nonce.n.md5_1[0], &b_nonce2.n.md5_1[0], 16)) // here where condition is getting failed and it is returning 2. Regards Surendra Hello, are there any error messages in the syslog? Maybe setting debug level for auth modules to 3 and looking at logs will help to reveal what is happening. Be sure syslog is in async mode, otherwise it will slow down processing. Cheers, Daniel On 16/10/15 08:42, Surendra.P wrote: > > Hello Daniel, > > > > I have a concern regarding auth module, I am running > 2000 TPS for 1000000 subscribers, using SIPP. What I observed when > you run small load like 500 TPS, everything get passed, but for 2000 TPS > > With same subscriber.csv file some of the subscribers getting > challenged. May I this why auth module is behaving like this.. below > are the parameters which I configured in auth module > > > > modparam("auth", "nonce_expire", 3600), rest all parameters default > values. > > > > Regards > > Surendra >

1 0

Changing a user's password in the Kamailio routing logic
by Michael McVady 16 Oct '15

16 Oct '15

Hi all, I was wondering if there was a way to change a user's password from within the routing logic section of kamailio.cfg (i.e. like do what 'kamctl password' does, but in the routing logic)? Thank, Mike

1 0

Multiple crashes of Kamailio 4.2.1
by Igor Potjevlesch 16 Oct '15

16 Oct '15

Hello, 3 crashes occurred today (we never seen crashes on this server before) during a period of 20 minutes. I had a look to the coredumps. Here is a quick overview of the first entries of "bt full": core.27671 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27656 : #0 0x00007f557700a048 in free_cell (dead_cell=0x7f5560abbce0) at h_table.c:175 core.27685 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27689 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27691 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.31977 : #0 0x00007f038e66fdee in free_cell (dead_cell=0x7f037811ebe0) at h_table.c:157 core.31994 : #0 0x00007f038e6ab65a in t_forward_nonack (t=0x7f037811ebe0, p_msg=0x7f038fdca438, proxy=0x0, proto=0) at t_fwd.c:1762 core.3700 : #0 0x000000000061697a in fm_insert_free (qm=0x7f19861fa000, frag=0x7f19864d53e8) at mem/f_malloc.c:245 core.3727 : #0 0x00000000006167a4 in fm_extract_free (qm=0x7f19861fa000, frag=0x7f19864f4cc8) at mem/f_malloc.c:208 I don't know how to proceed now. Can you help me to highlight the issue here? Thank you. Regards, Igor.

3 13

Re: [SR-Users] Auth module Issue:UA getting challanged thrice when load is high
by Daniel-Constantin Mierla 16 Oct '15

16 Oct '15

Hello, are there any error messages in the syslog? Maybe setting debug level for auth modules to 3 and looking at logs will help to reveal what is happening. Be sure syslog is in async mode, otherwise it will slow down processing. Cheers, Daniel On 16/10/15 08:42, Surendra.P wrote: > > Hello Daniel, > > > > I have a concern regarding auth module, I am running > 2000 TPS for 1000000 subscribers, using SIPP. What I observed when > you run small load like 500 TPS, everything get passed, but for 2000 TPS > > With same subscriber.csv file some of the subscribers getting > challenged. May I this why auth module is behaving like this.. below > are the parameters which I configured in auth module > > > > modparam("auth", "nonce_expire", 3600), rest all parameters default > values. > > > > Regards > > Surendra > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

routing based on TLS auth
by Max 16 Oct '15

16 Oct '15

Hi. I've got bunch of sip gateways to use and bunch of asterisk instances connecting to my Kamailio over TLS (own CA used). I'd like to match source with destination based on TLS authentication: * all the calls from asterisk A (TLS cert A) allowed via provider A * all the calls from asterisk B (TLS cert B) allowed via provider B ... * all the incoming calls from provider A forwarded to asterisk A (TLS cert A) * all the incoming calls from provider B forwarded to asterisk B (TLS cert B) ... * calls not coming from provider X or via TLS dropped. Providers are regular VoIP gateways (login:password, connection via SIP). I'm pretty sure this is possible with Kamailio but I'd appreciate any examples and pointers to modules I should use to implement this. Or detailed explanation of why this is a bad idea. Also, I'm not sure where it's better to place user database - should it be separate on per-asterisk basis? Central in Kamailio? No user db needed at all? cheers, Max.

2 1

Auth module Issue:UA getting challanged thrice when load is high
by Surendra.P 16 Oct '15

16 Oct '15

Hello Daniel, I have a concern regarding auth module, I am running 2000 TPS for 1000000 subscribers, using SIPP. What I observed when you run small load like 500 TPS, everything get passed, but for 2000 TPS With same subscriber.csv file some of the subscribers getting challenged. May I this why auth module is behaving like this.. below are the parameters which I configured in auth module modparam("auth", "nonce_expire", 3600), rest all parameters default values. Regards Surendra

1 0

Strange TLS problem
by Pete Kelly 14 Oct '15

14 Oct '15

Hi I am trying to use the [client] directives in tls.cfg for the first time. For my configuration I need to connect to 2 servers, both via TLS and both using different certs. I have defined an entry in tls.cfg like [client:1.2.3.4:5061] [client:5.6.7.8:5071] and I also have an entry in dispatcher tables for each server using the URI format sip:1.2.3.4:5061;transport=tls With this configuration, I expect dispatcher to send OPTIONS to each server, and Kamailio to resolve the host in dispatcher to the correct client section of tls.cfg and negotiate a connection. However this is not happening. Dispatcher module is trying to send OPTIONS as expected, and it is using TLS as expected but it is not using the correct [client] section from tls.cfg. Instead it is always falling back to [client:default], and if that is not present it then tries to look for some internal Kamailio defaults. Has anyone come across this issue before? Debug logs don't yield anything useful, however my suspicion is that Kamailio may be trying to match "sip:1.2.3.4:5061;transport=tls" from dispatcher module to "1.2.3.4:5061" from tls.cfg. Any help or advice would be appreciated Regards Pete

2 2

setflag and hash table variables
by Asgaroth 14 Oct '15

14 Oct '15

I originally sent this to sr-dev in error, I am resending to sr-users instead, thanks Hi All, Quick question, is it possible to use variables in the setflag function, I am trying to set a flag that is stored in a hash table using the setflag function, but I keep getting syntax errors. I am using kamailio 4.3.3. Here's what I am trying to do: htable core_setid defines a dispatcher setid key and its value is associated with a flag as follows: id:key:key_type:value_type:value:expires 1:102:0:1:7:0 so, simple hash table type with an integer key of value 7 and no expiry In dispatcher I have the following modparams set, so that ds_is_from_list() will set them if found: modparam("dispatcher", "setid_pvname", "$var(ds_setid)") modparam("dispatcher", "attrs_pvname", "$var(ds_setid_attrs)") Then in my routing logic I have the following code block: if ( ds_is_from_list() ) { if ( $sht(core_setid=>$var(ds_setid)) != $null ) { setflag( $sht(core_setid=>$var(ds_setid)) ); return; } else { send_reply("403", "No flag set for source"); exit; } } When I test this using kamailio -c -f <filename> I get the following error: 0(20629) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 21-52: syntax error 0(20629) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 21-52: missing '(' or ')'? 0(20629) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 21-52: bad command: missing ';'? 0(20629) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 53: bad command ERROR: bad config file (4 errors) If I change the setflag line to setflag(7), then I dont get the error. I was wondering if its possible to have hashtable variables as parameters to the setflag function, or am I going about this in the wron way. Any thoughts would be appreciated. Thanks

2 3

debian jessie and dependencies
by nadie 14 Oct '15

14 Oct '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello. If you install mariadb instead mysql Oracle, and then, you want to install kamailio with kamailio-mysql-modules: No se satisfacen las dependencias de los siguientes paquetes: mariadb-client-core-10.0 : Entra en conflicto: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. Rompe: virtual-mysql-client-core que es un paquete virtual. mysql-client-5.5 : Rompe: virtual-mysql-client que es un paquete virtual. Rompe: virtual-mysql-client-core que es un paquete virtual. mariadb-server-core-10.0 : Rompe: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. mariadb-client-10.0 : Rompe: mysql-client pero se va a instalar 5.5.44-0+deb8u1. Rompe: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. Rompe: virtual-mysql-client que es un paquete virtual. mariadb-server-10.0 : Rompe: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. Las acciones siguientes resolverán estas dependencias Eliminar los paquetes siguientes: 1) mariadb-client-10.0 2) mariadb-client-core-10.0 3) mariadb-server 4) mariadb-server-10.0 5) mariadb-server-core-10.0 Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJWGzvRAAoJEG5DH1ilMeeeLl4P/iLDNNIhW73Sducag6IuJaM6 VWeKhCpvkEGxUUiAfL5Byktme058W6d1y7KDUyBzzMLZysNBxm4kJAX4fZjOQBpO 6e1np/qYXC/1A54hAKrjynPIMFpofCG3W6I2wCGp9nKHhq/ail++SW7aFqht6nus yF9h/0mJjv8wNkZuzf0BGuZdTJZUqVBbAOoWNNZHVfqQ20IjbAL2rgg/ahXLt8tv BoYVBnCQAAActmiA2ETF13R0ioJY5gLUorn2IW/sq/TQyXiABaskmxY68d/z2svc DmIB3JaCpOOMeccU4Ay2A6zmo+HuN5LcMuPu0b+sk3aHSWn6Wl4JWGozjdneiNpp +ksNTJYhJm6dCvhsj4RIw+ALf0/mDF14YV+n0z6OiaG5V1pEt7Z7nGyVBY3uDHR3 Xm9oYkj3CKYW5nOgpimHir9f+1qD457AWMRkHyIJrOXbth0lSodUg+KbAw1mUci5 D1E5bf47M7+CbmAi0t7uOSKh2J+oQq4pW3SzZYQbrnWQnIkwpmFy72Oqp7GBiSD7 rxxKUt0xBIVjnzGQzCgQdLfeYJt3sZ8V9cHvnEe7Tte2hLRLdnqskOhmMkGxNavJ ZKA7McTjWFyXJNJAJAowtBLHbarIxo5oREAma1nJ42xLv9KyIhXpVao472lVJZLz V9kMzZksd4yhvAHH2eyi =oZca -----END PGP SIGNATURE-----

2 4

Kamailio vs RTPEngine Performance
by M S 13 Oct '15

13 Oct '15

Hi, I was asked some interesting questions yesterday by a rather techy client related to Kamailio and RTPEngine performance. Q1: For Kamailio we have say 8 child processes, which means there are 8 threads per listen address to process sip messages, i.e. for example 8 INVITE messages at any instance of time can be processed by Kamailio. If say, one INVITE takes 200ms to be processed and send out to destination, then we can estimate processing power of kamailio to be 40 INVITE requests per second per listen address, right? Q2: If so, then assuming each INVITE requires to engage RTPEngine. How can we ensure that RTPEngine would be capable to handle these 40 INVITE requests? The only relevant parameter I see in RTPEngine daemon script is "NUM_THREADS" (default value 5). What does this represent? Does this analogs to children as in Kamailio? Q3: How does RTPEngine manages SRTP <=> RTP translation (e.g. if one endpoint is WebRTC and other is traditional SIP)? My understanding is that RTPEngine has a kernel module (assuming kernel module is installed) which also manages this conversion besides forwarding the media packets. The Linux kernel already has encoders / decoders for nearly all encryption algorithms which are utilized by RTPEngine for doing the conversion in kernel space, right? if not then how it is done? Q4: Continuing to Q2, If these NUM_THREADS process actual media packets (RTP or SRTP) then are these synchronous or asynchronous? I think these are asynchronous, just want to confirm (otherwise RTPEngine won't process more then e.g. 5 calls at a time). So, assuming asynchronous how many packets can be queue to each thread? This would help estimating RTPEngine throughput using various codecs in calls. (e.g. assuming G.711 codec we have 50pps, if each thread queue size is 1000 then each thread can process 1000/50=20 concurrent calls and whole RTPEngine would process 20x5=100 concurrent calls). Thank you.

1 0

Capturing Kamailio retransmissions in routing script
by Olle E. Johansson 13 Oct '15

13 Oct '15

Hi! Is there a simple way to log outbound retransmissions by tm in the routing script? I would like to be able to see retransmissions. /O

2 1

evapi tcp parsing??
by Jayesh Nambiar 13 Oct '15

13 Oct '15

Hello, I'm exploring the evapi module for my kamailio to interface with an external node.js app for third party stuff like AAA, billing engine tasks, notifications and so on. I followed and took some ideas from the rtjson and evapi tutorial found here( http://kb.asipto.com/kamailio:k43-async-sip-routing-nodejs) to build the node.js app consuming events. When I stress tested the scenario using SIPp and tried sending a lot of events at 300-350cps from Kamailio, I noticed that at times the client is receiving 2-3 events in a single message together although I do event_sync_relay once per SIP message received and have netstrings enabled. I believe this is a typical behavior of TCP and needs to be handled by the client using some kind of Netstring handler. Please correct me if I'm wrong. And hence I'd like to know what particularly needs to be taken care of while writing a client that is listening for events on raw tcp socket and how does kamailio handle this situation while receiving messages over TCP socket?? Does kamailio recognize the end of netstring properly on evapi:message-received and give exactly one message to take care of on every "message-received" event or should that be handled in the script somewhere !! I also referred cgrates client over evapi example which is written in GO, but I couldnt find them handling TCP streams clearly either. I'd really appreciate some expert suggestion here to make an informed decision on using the evapi module for a large scale solution. Thanks, - Jayesh

3 26

How to get runtime configuration?
by Magnus Nordström 13 Oct '15

13 Oct '15

Hi, Is there any command to get the runtime configuration from kamailio? I tried "kamctl monitor" but there was not a lot of information. For example I need information about what tcp_conn_wq_max is set to. Best regards, Magnus Nordström

3 2

Re: [SR-Users] Lost 180 Ringing Response
by John Billings 13 Oct '15

13 Oct '15

> Hello, > > look at: > > http://kamailio.org/docs/modules/stable/modules/tm.html#tm.p.cancel_b_method > > Maybe one or another value for that parameter is suiting better your use > case. Thank you for the suggestion. This looked promising, but the cancel_b_method parameter is already at the default value of one. I also tried changing the value to two and got a similar result. This may sound crazy, but it almost seems like the problem happens only when kamailio is processing unrelated INVITES at the same time.

2 1

Re: [SR-Users] MySQL Connect Errors
by John Billings 13 Oct '15

13 Oct '15

2 1

Persistence of enable/disable for uacreg
by DanB 13 Oct '15

13 Oct '15

Hey Guys, I was wondering how does enable/disable flag for uac registration gets persistence, since I could not spot it in the mysql tables. Is that just temporary in-memory and on next restart/refresh does not longer take effect? Thanks, DanB

2 2

rfc: event route when kamailio starts
by Daniel-Constantin Mierla 12 Oct '15

12 Oct '15

Hello, based on a recent discussion with Alex Balashov (at Kazoocon this week), I pushed a patch that enables the execution of event_route[core:worker-one-init] when kamailio is started. It would be more like an alternative for event_route[htable:mod-init], for the cases when htable is not used, but also should ensure that all the modules are initialized. Moreover, because most of the modules close connection in mod init, I though of doing it when child one (udp receiver) initialization for all modules is finished, which means that all is ready for runtime, still this is before receiving any packet. On the other hand, forking is done by main process, so other children can start listening for sip. More or less what I want to ask here is whether the forking of the other children should wait until the event_route[core:worker-one-init] is finished (e.g., what is done in this event route is important before any sip packet is routed) or just leave it like it is. Worth to say that some modules do special processing in child init of worker one, like usrloc module loads the records from database to memory, which means that overall start of kamailio could be slower. Also, if there are time consuming actions in the event_route[core:worker-one-init], getting to the stage of routing sip packets will take longer. What would you find important, fork the other processes so routing sip will happen as soon as possible, or block until execution ofevent_route[core:worker-one-init] is finished. I guess that making it cusomizable in some way would be a solution, but knowing what use cases would be important may help in reducing the complexity to get it enabled. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

3 2

Dialog keep alive options are with error parsing headers
by José Seabra 12 Oct '15

12 Oct '15

Hello there, I'm using dialog module to send keep alive options to callee and caller in order to detect if dialog still up, but when kamailio generates the OPTION and parse it, gives the following errors: [root@c4-compute02-srv1 kamailio]# 37(1587) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg 0x7f89134c1d40 with 1 -> 3 37(1587) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2210 found on entry 2350 37(1587) DEBUG: dialog [dlg_req_within.c:427]: dlg_send_ka(): sending OPTIONS to caller 37(1587) DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=< sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [dns_cache.c:3293]: dns_srv_sip_resolve(): (62.28.197.25, 0, 0), ip, ret=0 37(1587) DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 14692 37(1587) DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: 37(1587) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> 37(1587) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK4693.2af96e95000000000000000000000000.0>; state=16 37(1587) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 37(1587) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 37(1587) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=90d40279ee5745d8851aa989e6fadfc2 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [64]; uri=[sip:unknown@62.28.197.6] 37(1587) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:unknown@62.28.197.6>] 37(1587) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <0> <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 *37(1587) ERROR: <core> [parser/msg_parser.c:96]: get_hdr_field(): ERROR: get_hdr_field: bad header* 37(1587) DEBUG: <core> [parser/msg_parser.c:280]: get_hdr_field(): get_hdr_field: error exit *37(1587) INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [VC* *]* 37(1587) DEBUG: topoh [topoh_mod.c:232]: th_prepare_msg(): parsing headers failed37(1587) DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: 37(1587) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> 37(1587) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK4693.2af96e95000000000000000000000000.0>; state=16 37(1587) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 37(1587) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 37(1587) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=90d40279ee5745d8851aa989e6fadfc2 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [64]; uri=[sip:unknown@62.28.197.6] 37(1587) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:unknown@62.28.197.6>] 37(1587) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <0> <OPTIONS> 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=e34d43e164ad4b7392dc9c6e1058a983 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: dialog [dlg_cseq.c:215]: dlg_cseq_msg_sent(): traking cseq updates 37(1587) DEBUG: dialog [dlg_hash.c:671]: internal_get_dlg(): ref dlg 0x7f89134c1d40 with 1 -> 4 37(1587) DEBUG: dialog [dlg_hash.c:674]: internal_get_dlg(): dialog callid='13208317017c42128fbf00e1685ee1b3' found on entry 2350, dir=2 37(1587) DEBUG: dialog [dlg_cseq.c:233]: dlg_cseq_msg_sent(): request not going downstream (2) 37(1587) DEBUG: dialog [dlg_hash.c:846]: dlg_unref(): unref dlg 0x7f89134c1d40 with 1 -> 3 37(1587) DEBUG: dialog [dlg_req_within.c:452]: dlg_send_ka(): keep-alive sent to caller 37(1587) DEBUG: dialog [dlg_req_within.c:427]: dlg_send_ka(): sending OPTIONS to callee I'm using the following dialog configuration: modparam("dialog","db_url", DBURL) modparam("dialog","db_mode", 0) modparam("dialog","enable_stats", 1) modparam("dialog","dlg_flag", DLG_FLAG) modparam("dialog","profiles_with_value","concurrentcalls") modparam("dialog", "detect_spirals", 1) modparam("dialog","dlg_match_mode", 2) modparam("dialog", "initial_cbs_inscript", 0) modparam("dialog", "default_timeout", 3600) modparam("dialog", "send_bye", 1) modparam("dialog", "timeout_avp", "$avp(s:timeout_dlg)") modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) I'm also using topoh module. Anyone can help me to identify if I'm doing the proper configuration? Thank you for your support Best Regards José Seabra

3 9

Startup Route in Kamailio
by Sunil More 12 Oct '15

12 Oct '15

Hi All, I am new to Kamailio world. I am trying to initialize some details at Kamailio startup, this requires some database queries from a few tables. I have read about the htable event, however sql queries are not working in this route. I have come upon following errors. Please suggest a solution for this. Route used: event_route[htable:mod-init] { xlog("L_INFO", "********Initailising the startup route ******\n"); sql_xquery("cb","SELECT id, media_server_port, name FROM domains","ra"); } error : Oct 8 13:28:32 P172 /usr/local/kamailio/sbin/kamailio[28735]: INFO: <script>: ********Initailising the startup route ****** Oct 8 13:28:32 P172 kamailio: ERROR: <core> [daemonize.c:303]: daemonize(): Main process exited before writing to pipe Thanking You, Sunil More Ph : 9503338275

2 1

Familiar topic about Kamailio with Asterisk behind NAT
by Dirk Teurlings - SIGNET B.V. 09 Oct '15

09 Oct '15

Hi, I found some existing topics on this but failed to get a solutions out of them. We're running into some issues with client devices connecting to our private addresses. The way it is setup now: CLIENTS <-> (NAT) <-> INTERNET <-> KAMAILIO(4.2.5) with RTPPROXY(v1) <-> PRIVATE LAN <-> ASTERISK (v1.8) Our Kamaialio and Asterisk are in a private address range, but Kamailio also has a public interface. Most of the clients (about 95%) work well with this setup, but a couple don't. We have one case now where the CLIENT tries to connect to the private address of ASTERISK. And of course, that doesn't work. I'm kind of stuck as to where I need to fix this. I tried using the externaddr option in Asterisk to solve it on that end. But that didn't help anything. The NAT options in Kamailio are not really suited for this, as they tend to fix client NAT problems. Any pointer or help would be greatly appriciated. Cheers, Dirk

8 16

[pua] Question to DB Mode 0
by Maciej Marczyński 09 Oct '15

09 Oct '15

Hi Guys, I have a question regarding pua module parameter 'db mode'. I run Kamailio (4.3.2) with sqlite and I tried to set the parameter to 0 but even then I received: > ERROR: db_sqlite [dbase.c:489]: db_sqlite_commit(): sqlite commit failed: NOT NULL constraint failed: pua.call_id > ERROR: db_sqlite [dbase.c:138]: db_sqlite_cleanup_query(): finalize failed: NOT NULL constraint failed: pua.call_id > ERROR: pua [pua.c:1179]: db_update(): while inserting in db table pua > > As far as I checked the documentation, it says that '0' means: > high speed memory based storage Does it still writes to DB anyway (on timer or whatever?) or just keep it in memory? Is there any way to disable writing to DB? The second thing is the reason of the ERRORs above. Why does pua think that my call_id is null? If you need any more information please let me know. Best regards and have a good weekend, Maciej

3 7

Siremis debug window
by Infinicalls Infinicalls 08 Oct '15

08 Oct '15

Hi, I have installed Siremis and modified it for public user registration. I have added a new user. When I try to log in it opens up a new (debug) window and prints "1". It doesn't allow me to log in. But I can log in as admin;admin. And siremis/user/register doesn't load properly. It redirects to the login page again. Kindly let me know how do I solve this. regards Ganesh Kumar

2 8

MySQL Connect Errors
by John Billings 08 Oct '15

08 Oct '15

We are using kamailio 4.2.0 with auth_db enabled and connected to a MySQL server. Occasionally, in the middle of the night when there is no activity we get the following errors out of the blue (the last three lines are repeated indefinitely). According to the MySQL site <http://dev.mysql.com/doc/refman/5.0/en/error-lost-connection.html>, this can be caused by a connect timeout that is too short (We verified that Aborted_connects is steadily rising). The kamailio server is right next to the database server so there is no network latency. We increased the db_mysql timeout_interval parameter, but we still see MySQL’s Aborted_connects counter incrementing. So it’s like kamailio is still timing out on connection attempts. Ideas? Thanks. Oct 1 02:34:49 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:34:49 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:34:55 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:34:55 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:34:55 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Host ‘X.X.X.X' is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts' Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Host 'X.X.X.X' is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts' ...

2 1

TM not ending transactions
by Daniel Tryba 08 Oct '15

08 Oct '15

Running 4.3.x (both .1 and .2) I have seen it happening twice sofar that dialogs (the module) will not end, resulting in to calls failing due to call limits based on active dialogs. Accounting logged a BYE for all these calls. kamcmd tm.stats shows an increasing number of current/waiting transactions (as grphed in attachment). Has anyone seen this problem?

3 7

Questions about LCR
by Scott C. Fertig 08 Oct '15

08 Oct '15

Hello, Sorry to bother and if I sound like a newb thats because I am kind of new (but im trying =) ) I am having some issues getting started using LCR and was wondering if anyone could point me in the right direction and/or let me know what I am doing wrong. I began setting up the other day using kamailio (4.4.0-dev5) from git (maybe I picked the wrong version?) and am struggling to get lcr to work properly. So far I have been able to get the module to load, added the route to the config and have added some LCR values. Inbound calling is working perfectly to a registered device which is great so far. This is my config for the LCR: http://pastebin.com/LUF0Pgr1 I have the following values in my database for LCR: http://pastebin.com/bDpQug0J I was not sure if dumping all the text in the email would be appropriate if needed please let me know to resend with the configs in the email. Whats strange to me is that even with the xlog I do not get any messages to syslog at all, I have defined WITH_DEBUG, debug level at 3, and log facility set to LOG_LOCAL0 (Sorry if I'm bringing up multiple issues) Anyways back on point, I'm wondering if this is related to the avp's listed, I am not sure what the avps referenced are supposed to represent, when I try to dump all the avps I get a empty list, and it is querying the usr_preference table for these. Is there something I need to add within this table, or is there perhaps something wrong with my configuration? [root@localhost kamailio]# kamctl avp list database engine 'MYSQL' loaded Control engine 'FIFO' loaded Query: SELECT uuid,username,domain,attribute,type,value FROM usr_preferences ; Dumping AVPs Thank you!

2 2

Lost 180 Ringing Response
by John Billings 07 Oct '15

07 Oct '15

Hi Everyone, We are using Kamailio 4.2.0 with TLS. It’s setup so multiple phones can register to the same account and they all ring together. It always works great unless one of the phones is on a slower Internet connection. Occasionally that phone will start ringing after another phone has already picked up the call. The problem is that the INVITE to the slow phone is not followed by a CANCEL. So the phone keeps on ringing for a long time. When there is no problem, the logs on the phone show: Receive: INVITE Send: 180 Ringing Received: CANCEL Send: 200 Ok When the phone keeps ringing: Delay: up to several seconds Receive: INVITE Send: 180 Ringing The problem never seems to happen if I wait until all of the phones have received the INVITE. So is the late 180 response not getting handled by the right session? Any ideas on how to track this down?

2 1

Re: [SR-Users] uac_req_send and reply timeout
by Paweł Sternal 07 Oct '15

07 Oct '15

On 07.10.2015 16:41, Daniel-Constantin Mierla wrote: > Hello, > > in the event route for uac_req_send() is practically during handling the > reply for what was sent before. Also, the uac_req_send() sends something > out of the processed sip message, so typically functions won't work > because they will be applied to the processed sip message. > > Maybe try setting the avp with the timeout value (see tm parameters), > might work, I haven't checked -- report the results, would be good to know. Unfortunately solution with fr_timer_avp also doesn't work. Only set globally fr_timer works. Well... there is always a solution. Thanks anyway. > Cheers, > Daniel Regards, Paweł

2 1

uac_req_send and reply timeout
by Paweł Sternal 07 Oct '15

07 Oct '15

Hi, It is posibility to set timeout for uac_req_send with setting "evroute" to 1. Yes, I found a solution without this set when using event_route[tm:local-request] [1] but with "evroute" option this don't work ofcorse. regards Pawel [1] http://lists.sip-router.org/pipermail/sr-users/2011-October/070695.html

2 1

SIREMIS Help
by Vojimir Raovic 07 Oct '15

07 Oct '15

Hi, Just wanted to see how to get SIREMIS user manaual? Regards, Voyimir [image: me]*Vojimir Raovic*, Principal Engineer & Co-founder *Office:* +61 8 9200 3500 | *Web:* www.norwoodsystems.com | *Twitter: * @norwoodsystems <http://twitter.com/norwoodsystems> *Direct:* +61 8 9200 3503 | *Mobile:* +61 450 019 775 [image: me][image: me] [image: me] NOTICE: If you are not an authorised recipient of this e-mail, please contact Norwood Systems immediately by return e-mail or by telephone on +61 8 9200 3500. In this case, you should not read, print, re-transmit, store or act in reliance on this e-mail or any attachments, and should destroy all copies of them. This e-mail and any attachments are confidential and may contain legally privileged information and/or copyright material of Norwood Systems or third parties. You should only re-transmit, distribute or commercialise the material if you are authorised to do so. Any legal privilege attaching to the contents of this e-mail or any attachments is expressly reserved and is not waived by any transmission to an unauthorised recipient. This notice should not be removed. *Please consider the environment before printing this e-mail.*

2 1

Recall: Kamailio B2BUA or Dual router ?
by Mark Henriksen 07 Oct '15

07 Oct '15

Mark Henriksen would like to recall the message, "[SR-Users] Kamailio B2BUA or Dual router ?".

1 0

Kamailio B2BUA or Dual router ?
by Mark Henriksen 07 Oct '15

07 Oct '15

Hello All I have been looking for answers and trying really hard to figure out if I can use Kamailio as a Dual Router (Inbound and Outbound using 2 IPs on 1 Eth "Amazon AWS") I can not seem to find information is this is possible using Kamailio or if am just plain "stupid" and can not read the Wiki well enough J Hope you can help //Mark

2 2

Kamailio as a LB with Asterisk and consul
by Sylvain Boily 06 Oct '15

06 Oct '15

Hello, I did a new small project with Asterisk to register it on consul (http://consul.io) to use a discovery service tools. The project is here : https://github.com/sboily/asterisk-consul-module This last day, i'm working to use kamailio as a loadbalancer for Asterisk and autoscale it. You can find example on my docker-compose.yml, contribs directory and README. It's not perfect, because i'm clearly not a kamailio expert, but it works. So, i'm looking about feeling to this dev, and i would expect some help from kamailio community to configure kamailio as a LB better ;) I will be at astricon, so if people want to share with me about this project, contact me. Have a good day Sylvain

2 2

creating debian package from source
by Thibault Gueslin 06 Oct '15

06 Oct '15

I have difficulties to install the debian package of kamailio I have build from source Creation works fine. (make cfg, configuration of proper debian in pkg/debian/deb , make deb) But kamailio tries to find module here : /usr/lib/x86_64-linux-gnu/kamailio/modules/ and modules are installed in /usr/lib64/ Of course, it works fine by specifying -L with actual path. I have tried also to download the source of debian package with same result. I have missed some steps. Thibault

1 0

Http authentification with JSON-RPC command
by Loïc Boissy 06 Oct '15

06 Oct '15

Hello, I would like to authorize JSON-RPC commands for a specific user. Is it possible to enable HTTP authentification on the xHTTP module? Thanks for your help. Best regards Loic BOISSY

3 3

Re: [SR-Users] diameter auth with round trip
by JB 06 Oct '15

06 Oct '15

Thanks for the suggestion about auth_diameter, do you think there would be an interest in the community if we were to do this modification on ims_auth or auth_diameter ? could it be integrated to trunk ? We do not want to publish our own fork of ims_auth or auth_diameter with that small change and maintain the code repository if it can be avoided, so it would be nice if we can put it in the trunk and forget about it. Plus it is more open source oriented to publish to main branch i feel :) JB 2015-10-03 23:53 GMT+02:00 JB <jbf.nospam(a)gmail.com>: > Hello all, we are working on a SIP solution using Kamailio. > > We want to secure our base of user credentials even in case of attack on > the SIP server, and for that reason we plan to use diameter authentication > as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt > > Paragraph 6.2 describes a mode where the HSS answer with code > > DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. > > > This does NOT corresponds to what is done on Kamailio module ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) > > ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) > > Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? > > Thank you > > JB > >

2 1

diameter auth with round trip
by JB 06 Oct '15

06 Oct '15

Hello all, we are working on a SIP solution using Kamailio. We want to secure our base of user credentials even in case of attack on the SIP server, and for that reason we plan to use diameter authentication as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt Paragraph 6.2 describes a mode where the HSS answer with code DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. This does NOT corresponds to what is done on Kamailio module ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? Thank you JB

2 1

Prepend values to PSTN gateway?
by Infinicalls Infinicalls 05 Oct '15

05 Oct '15

Hi, I am able to successfully connect to a PSTN gw from Kamailio. Now, he says I need to prepend a value (eg.123123) before the number(ENUM) and then only terminate the calls. Where do I add this? Can somebody help me? thanks Ganesh Kumar

3 3

Interoperability issues between browsers and mobile phones to place a call.
by Safdar Khan 05 Oct '15

05 Oct '15

Hi Daniel sir I have enabled web-sockets module into my kamailio sip server.I am trying to make call from web browsers and mobile phones. To make a call from web browser i am using sipml5 and as a mobile app i am using imsdroid. However i have done some search on net and i found nothing relevant. Any suggestions and help?.Waiting your response eagerly. Thank you

2 3

defaults missing from tcp ops module readme
by Juha Heinanen 05 Oct '15

05 Oct '15

after seeing some commits on tcpops module, i decided to check its readme and noticed that nothing is mentioned about the default values of parameters. it might be useful to tell that, e.g., if tcp_keepalive_enable is not set, then tcp_keepalive core parameter value tells what the status is. -- juha

2 2

problem with dispatcher load balancing
by Ding Ma 05 Oct '15

05 Oct '15

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"); }

2 10

Changed the bindport of Kamailio but still listening to older port
by amjad ali 04 Oct '15

04 Oct '15

Hi, I am scratching my head for one month to get this sorted and I have spent so many hours but couldn't figure out. Please help ! I have changed the bind port to 5099 but it is still listening to the previous port even I have rebooted the system. My Code is : kamailio.bindip = "192.168.1.105" desc "Kamailio IP Address"kamailio.bindport = "5099" desc "kamailio port" However, when I restart kamailio service it shows that it is still listening to 5060 (attached the screen shot) Regards,Amjad

2 1

Kamailio is not forwarding the call to asterisk
by amjad ali 04 Oct '15

04 Oct '15

Hi, I have kamailio and asterisk running on same machine. When I make an internal call, it routes the call to other extension. However, if I stop the asterisk service then the call still routes the same way to other extension. Is there a chance that kamailio does the call routing it self. Also, I cannot see any sip extension registered at asterisk when I run a command "Sip show peers" neither there is any activity in Asterisk's log. Regards,Amjad

2 1

Inconsistent string interpolation
by Sven Neuhaus 02 Oct '15

02 Oct '15

Hello, I'm occasionally running into issues with inconsistent string interpolation in the kamailio.cfg. Here's the latest example. Take this line: $var(x) = "sip:+$var(abc)-$var(def)-$si@$Ri:$Rp"; Here, the values of the variables will NOT be interpolated into the string. Instead, I have to use this: $var(x) = "sip:+" + $var(abc) + "-" + $var(def) + "-" + $si + "@" + $Ri + ":" + $Rp; However, when I have an instruction such as: append_hf("P-Asserted-Identity: $var(x)\r\n"); Then string interpolation takes place (i.e. the value of $var(x) is put into the string. This is very confusing and error prone. Is it something that can be fixed? The behaviour should be consistent. -Sven

2 6

Kamailio v4.3.3 Released
by Daniel-Constantin Mierla 02 Oct '15

02 Oct '15

Hello, Kamailio SIP Server v4.3.3 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.3. For more details about version 4.3.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/10/kamailio-v4-3-3-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

1 0

[ims_auth] compute authenticate field in diameter server, not kamailio
by JB 02 Oct '15

02 Oct '15

Hello all, we are working on a SIP solution using Kamailio. We want to secure our base of user credentials even in case of attack on the SIP server, and for that reason we plan to use diameter authentication as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt Paragraph 6.2 describes a mode where the HSS answer with code DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. This does NOT corresponds to what is done on Kamailio ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? Thank you JB

1 0

reSIProcate v1.10.0 released
by Daniel Pocock 02 Oct '15

02 Oct '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 The v1.10.0 release has been tagged and made available for download: https://www.resiprocate.org/files/pub/reSIProcate/releases/ and a direct link to the ChangeLog: https://github.com/resiprocate/resiprocate/blob/resiprocate-1.10/ChangeLog Debian, Fedora and Ubuntu packages are in the process of being updated. It is about 20 months since 1.9.0 and there have been many improvements. Highlights of the v1.10.0 release: - - repro SIP proxy now includes: - full presence server - PostgreSQL support - Include directive in the config file - - enhancements to the stack and proxy include: - WebSocket authentication may now use URL parameters instead of cookies, client certs or DIGEST - better Syslog support - TLS improvements (PFS, private key passphrases) - better assertion handling (resip_assert) - WSAPoll for Windows, allowing thousands of TCP connections - removing transports at runtime - added netns (Linux network namespace) support The RTC Quickstart Guide provides an overview of how to get started using repro, Kamailio and other popular SIP, XMPP and TURN products http://rtcquickstart.org Contributors to v1.10.0 include Alan Hawrylyshen, Alexander G. Pronchenkov, Byron Campen, Daniel Pocock, Daniel Tacalau, Dan Petrie, Dario Bozzali, Fedor Brunner, Gregor Jasny, Kevin Green, Reiner Herrmann, Robert Sparks, Scott Godin, Thomas Thorne, Vasil Kolev and Wolfgang Rosenauer. Please also see the hall of fame in Github: https://github.com/resiprocate/resiprocate/graphs/contributors which emphasizes the enormous and sustained contribution Scott Godin has made to this project and this release in particular. We are also very grateful to all those who contributed feedback through the mailing list, testing the beta packages in Debian and all past contributors to previous releases of the reSIProcate project. reSIProcate point releases are now announced on the low-volume resiprocate-announce mailing list, please subscribe if you would like to know when 1.10.1 comes along: http://list.resiprocate.org/mailman/listinfo/resiprocate-announce SHA256 checksums for the files: resiprocate-1.10.0.tar.gz c089686e183b8a739064731fa9e5002ab4d0bd6908cd8f5e648e021d76363c98 resiprocate-1.10.0.zip df4ba3902770390e7a0f9f6b13c5959292feba04785f01b9d044ad31325fa1e0 resiprocate-contrib-1.10.0.tar.gz 2f40c568644cbf1dd8628da0535dadd8d17367860234163d003fe0fbd75ba9a0 resiprocate-contrib-1.10.0.zip ef1c50b7c70b533c7bec1bfb071ca19ff9368a0b1e8a951e430e273b8d413715 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1 iQIcBAEBCAAGBQJWDlraAAoJEOm1uwJp1aqDCxgP/jlJBE0fTSp1FzohA5IRmrmQ rVQcKOB6pI6rWTMEwv8WuBs6sA7z/z94b2jbLYZRo/iwXXb85TbQXqnIQkyw5GAs l+F73B+rPyNYN46CZvjYv/rrscW8rxDPoein/t7uEzfATm1xgAZrIykWhZx42WV2 QUEjZDAXRm2Q2duIBfnxyTmhyFurWgHJhtxLzx6zCVnml/hWt8xslzkH1WYtfXuI 0ltGqNuVlYlhHCi2/BpaxHVdsPELhdiTHpIu88UgBSHI5gKiUBp38dhQttwJr+bt VWF9Ri+KQPbn7d5JGc8mPMeBTlYmJUgFY4s//eIU2pYRcxZ2w1/iim2SHewMpift 51C9Lmp6vSBycpCNPAQ6yQqJdAjIlTlY9eKJ2td+bCUw66Zlx10h3tqY4ka95o0s TfFporu6dpaIu3TSo+qVW8fhXuYmRh86NHLLi3+LCbZ9i6j5E71qor5mkz9x/xLM 7kWbVJKNJFTdbTsVjAKZ7uh4YFfkn3NcYEu+5zwF5Fm/z6ohE36sme12Lo0khjyq kTDhefATpFvZLxu2l0RBmG+NktOj/H2MJvbPVGPyEtHaam1BJaRPv4UMh6bldoR4 vsrt8cL31HhnWGTZS7M7/yg39Ge+JrTAHQD8wpdVh6nBzFTqb2n2WIk2swPICrLA v8lx2bBfol6NYvB0GhAT =R1HL -----END PGP SIGNATURE-----

1 0

Siremis-4.2.0 - error with Alias and login page
by Infinicalls Infinicalls 02 Oct '15

02 Oct '15

Hi, I tried installing siremis-4.2.0 following the step given.The Alias was not working. So, I installed it in the /html folder and the installation was completed. But the login page looked strange. There was no logo displayed and the submit button was not there. Mine is apache 4.2 on Ubuntu LTS 14. Kindly let me know how to solve this problem. thanks Ganesh Kumar

2 2

how to use data received from diameter answer(CCA) in kamailio config script
by SureshTummala 02 Oct '15

02 Oct '15

Hi, I want retrieve the data present in the Credit-Control-Answer response from diameter in kamailio configuration script. Is there any provision in kamailio 4.1.x (or) later? Any suggestions are appreciated. Regards,Suresh Tummala

4 3

FW: Your iptel.org Registration
by Bijal Desai 02 Oct '15

02 Oct '15

When I call on sip:bijaldesai@iptel.org, it is not working. Can you please guide me -----Original Message----- From: registrar(a)iptel.org [mailto:registrar@iptel.org] Sent: Tuesday, September 29, 2015 10:00 PM To: Bijal Desai <Bijal(a)ssminfotech.com> Subject: Your iptel.org Registration Thank you for registering with iptel.org. We are reserving the following SIP address for you: sip:bijaldesai@iptel.org To finalize your registration please check the following URL within 24 hours: http://serweb.iptel.org/user/reg/confirmation.php?nr=473f1e2fc74bfb3e9d28eb… (If you confirm later you will have to re-register.) Here are the details of your account: --------------------------------------------------------------------- Username: bijaldesai(a)iptel.org Password: Angel1987@ Email: bijal(a)ssminfotech.com Full Name: Bijal Desai Sip address: sip:bijaldesai@iptel.org (You can always retrieve these via the "Forgot Password" link on the signup page) -- This message is automatically generated by SerWeb.

2 1

Kamailio-Asterisk real-time integration with Dynamic IP
by Rudy Eschauzier 02 Oct '15

02 Oct '15

All, The Kamailio-Asterisk integration that is described here asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [Asipto - SIP and VoIP Knowledge Base Site] is limited to Asterisk servers with a fixed IP address. The reason is that Kamailio identifies and authorizes traffic from Asterisk based on the source IP address. Attached is a patch to kamailio.cfg that adds support for dynamic Asterisk hosts. It introduces the option to define a fully-qualified domain name instead of the IP address to identify the Asterisk server. The host name can be dynamically updated to always point at the (changing) IP address of the Asterisk server. Defining WITH_ASTERISK_FQDN enables the new option. When WITH_ASTERISK_FQDN is set, Kamailio will perform a DNS search on the host name in the asterisk.bindhost variable. It will authorize incoming traffic if it originates from one of the IP addresses the DNS lookup returns (and matches the asterisk.bindport variable as before). Furthermore, the patch introduce the option to automatically set the IP address of the server that Kamailio is running on. This does away with the need to properly define kamailio.bindip on every server that the configuration is running on. This option is enabled by defining WITH_AUTOIP. With the suggested changes I have been able to successfully run an Asterisk server in my home, having a dynamic public IP address and connecting to the Kamailio proxy on a VPS server with a static IP addres. Please let me know if you have any comments or suggestions. Kind regards,Rudy Eschauzier. | | | | | | | | | | | asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [...Table of Contents Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database Architecture Registration Call Initiation Requirements MySQL Insta... | | | | View on kb.asipto.com | Preview by Yahoo | | | | |

2 1

cnxcc module
by Максим Панчоха 02 Oct '15

02 Oct '15

Hello! I'm trying to set max call duration, but after 5 seconds call is continues. Kamailio version: 4.3.2 Here's part of my config file: #!define CC_FLAG 29 ... modparam("cnxcc", "credit_check_period", 1) modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) ... if ((is_method("INVITE")) && (is_request()) && (!has_totag())) { route(CNXCC); } route[CNXCC] { setflag(DLG_FLAG); if (!is_method("INVITE")) return; $var(client) = $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$var(client)", "$var(max_time)")) { xlog("Error setting up credit control"); return; } } event_route[dialog:failed] { xlog("L_INFO", "********END CALL*********$DLG_lifetime*******************"); } event_route[dialog:end] { xlog("L_INFO", "******END CALL************$DLG_lifetime****************"); } event_route[dialog:start] { xlog("L_INFO", "***********************DIALOG STARTED****************************"); } event_route[cnxcc:call-shutdown] { $var(duration) = $TS - $dlg(start_ts); xlog("L_INFO", *****END CALL***BALANCE IS EMPTY************$var(duration)*************"); } Tell me please, where I made a mistake. Thank you for a response.

2 1

ERROR: mediaproxy [mediaproxy.c:1978] #347
by Ngoc 02 Oct '15

02 Oct '15

Hello supporter, I took a look kamailio and get the error: Sep 29 11:28:37 lvps46-163-74-198 /usr/local/sbin/kamailio[1738]: ERROR: mediaproxy [mediaproxy.c:1978]: EngageMediaProxy(): engage_media_proxy requires the dialog module to be loaded and configured. How to resolved this problem, please help me. Regards, --------------------------- Dinh Hong Ngoc Technical Lead Edge-works Software Co.,ltd Hall 5, Quang Trung Software City, Tan Chanh Hiep Ward, Dist. 12, Ho Chi Minh City, Vietnam. Email: ngoc.dinh(a)edge-works.net Phone: +84 8 54371075 | Mobile: +84 909 911 406 Site: http://www.edge-works.net

2 1

receive_fd(): EOF
by jery john 02 Oct '15

02 Oct '15

11(12454) : <core> [mem/q_malloc.c:149]: qm_debug_frag(): BUG: qm_*: fragm. 0x7f9e8fcb2448 (address 0x7f9e8fcb2478) end overwritten(c0c0c000, abcdefed)! 13(12456) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 20 0(12443) ALERT: <core> [main.c:784]: handle_sigs(): child process 12454 exited by a signal 6 0(12443) ALERT: <core> [main.c:787]: handle_sigs(): core was generated 0(12443) INFO: <core> [main.c:799]: handle_sigs(): terminating due to SIGCHLD 13(12456) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 12(12455) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 9(12452) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 10(12453) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 8(12451) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 4(12447) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 3(12446) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 7(12450) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 6(12449) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 5(12448) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 2(12445) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 1(12444) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 0(12443) INFO: <core> [sctp_core.c:48]: sctp_core_destroy(): SCTP API not initialized using kamailio 4.2 with modified tcp_read.c and tcp_main.c files for buffer manipulation both files attached gdb output Reading symbols from /usr/local/sbin/kamailio...done. warning: exec file is newer than core file. [New LWP 15502] warning: Error reading shared library list entry at 0x79e8c789480056e4 warning: Error reading shared library list entry at 0x482a50b70fd8458b Core was generated by `/usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -P /var/run/ka'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000000000061f73d in _shm_resize (p=0x61f729 <_shm_resize+541>, s=32767, file=0x3f <error: Cannot access memory at address 0x3f>, func=0x0, line=-1433800800) at mem/shm_mem.c:111 111 DBG("WARNING:vqm_resize: resize(0) called\n");

2 1

blf problem
by huseyin kalyoncu 01 Oct '15

01 Oct '15

Hi I am trying to implement BLF functionality as described in http://kb.asipto.com/kamailio:presence:k43-blf. I am using the same conf with the one in the tutorial except that i am using postgres database instead of mysql. The problem is BLF is not working with this config (presence server and proxy server are in the same kamailio with same IP). But when I separate presence server from proxy server (2 different kamilio boxes with different IPs) BLF working perfect. i can see PUBLISH & NOTIFY messages in the tcpdump output with the working configuration. But i can not see any PUBLISH messages in the tcpdump output with the original config (one with presence and proxy servers are same). My question in the normal working environment (when BLF working perfectly); should i see PUBLISH and NOTIFY messages in the tcpdump output when the presence and proxy servers are the same? thanks Huseyin

2 9

SIP and conntrackd experience?
by Grant Bagdasarian 01 Oct '15

01 Oct '15

Hello, Does anyone have experience with SIP and conntrackd? Our media and signaling is separated. Does this work for VPN connections where SIP is sent over the VPN but the media over public internet? Thanks, Grant

2 5

modifying $fu / $fU (R-URI From)
by Sven Neuhaus 01 Oct '15

01 Oct '15

Hello, I'm having some troubles with modifying $fU in my kamailio.cfg. The following line appears to have no effect: $fU = $(fU{s.substr,2,0}); I would expect it to remove the first two leading characters from $fU. Has anyone else seen this? I have the same issue with modifying $fu. We're not using the uac module. And we're using the latest stable version of Kamailio. Thanks, -Sven

4 4

out of memory Error
by Yasin CANER 01 Oct '15

01 Oct '15

3 15

Crash kamailio 4.3.1
by Thibault Gueslin 30 Sep '15

30 Sep '15

After upgrading to kamailio 4.3.1 and modifying the config, I get this crash very often: ... CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 21 ALERT: <core> [main.c:728]: handle_sigs(): child process 9341 exited by a signal 11 ALERT: <core> [main.c:731]: handle_sigs(): core was not generated INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD INFO: <core> [main.c:794]: sig_usr(): signal 15 received INFO: <core> [main.c:794]: sig_usr(): signal 15 received INFO: <core> [main.c:794]: sig_usr(): signal 15 received ... Is it known ? or may be related to my config ? Thank you Thibault

3 23

db_text and caching/reloading
by Asgaroth 30 Sep '15

30 Sep '15

Hi, I am busy testing the db_text module for some modules who's configuration does not change that often. If I set the db_mode for db_text to 0 (caching), it appears that if I modify the file directly (in this case the address dbtext file) and then issue a "kamctl address reload" the updated contents are *not* reloaded. However, if I set db_mode to 1, then the contents of the db_text file automatically reload when I change it on the file system. Is it possible to run db_text in caching mode but to also force a reload if I manually update the underlying database file? Additionally, in kamctlrc, I had to add the following variable: DBTEXT_PATH="/etc/kamailio/db_text" Whereas the original example for db_text path in kamctlrc is as follows: # database path used by dbtext, db_berkeley or sqlite #DB_PATH="/etc/kamailio/db_text" Is this just a typo in the kamctlrc distribution or is the variable meant to be DB_PATH or DBTEXT_PATH? Thanks

1 3

ACK and BYE are forwarded statelessly (failed to parse Route HF)
by Miroslav Dunaev 30 Sep '15

30 Sep '15

Hello everyone, I have a setup where kamailio acts as UAC to a SIP provider. Kamailio is also configured to pass all incoming calls to an Asterisk. Here is a diagram: Phone<-->Asterisk(10.10.11.232)<--->(PRIV:10.10.11.238)Kamailio(PUB:44.44.44.148)<--->(194.213.29.92)SIP-provider. When a call comes in from the SIP-provider, it gets all the way to the phone. But when the SIP-provider sends BYE or ACK (during hungup event or 200OK/SDP asterisk response.). I get this errors in my kamailio log: siputils [checks.c:100]: has_totag(): totag found <core> [parser/parse_rr.c:49]: do_parse_rr_body(): parse_rr_body(): No body for record-route rr [loose.c:106]: find_first_route(): failed to parse Route HF rr [loose.c:929]: loose_route(): There is no Route HF tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=10101, isACK=1 tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed ..... .... .... siputils [checks.c:100]: has_totag(): totag found tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=2 , global msg id=1 , T on entrance=0xffffffffffffffff tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=10101, isACK=1 tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found tm [t_funcs.c:279]: t_relay_to(): SER: forwarding ACK statelessly here is the link to: kamailio.cfg, ngrep -i any capture, debug_level6 log: http://labmir.com/kamailio/ Looking at the SIP ACK that comes from SIP provider I can see that they use strict routing. (i.e. TO field is set to kamailio's public IP that is next hop to them). Maybe I am wrong to judge that its not loose routing just because of the TO field. Anyways, because of (failed to parse Route HF) error, ACKs and BYEs are forwarded by kamailio statelessly, as a result kamailio uses wrong interface to forward them (SIP_dailog.txt line:233) and therefore asterisk never sees ACKs or BYEs. How can I make kamailio recognize ACKs and BYEs belonging to a dialog and pass them statefully through the right interface? It is worth to mention that I use exactly the same config with a different SIP-provider (which is peered to our PUB IP not registered through UAC like in this case) and it works flawlessly. Thank you, Mir

2 1

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users