- sr-users - kamailio.org

wiresharking TLS problem
by Rex Lin (林昱頡) 27 Mar '17

27 Mar '17

hi all wondering if any of you has the experience of TLS decrypting, SIP client has problem while calling each other, but we are using TLS for communication protocol.....facing problem debugging it..... Best Regards, Rex Lin

3 3

Regex module having trouble with large regex expressions in kamailio 4.4.5
by Tim Balmer 24 Mar '17

24 Mar '17

Hi, I discovered some strange behavior from the regex module yesterday (kamailio version 4.4.5). For example we have the phone number 0033755500381 that we try to match with the following kamailio code: #!define REGEX_FF_NEG_INTERNATIONAL "6" ... if (pcre_match_group("$rU", REGEX_FF_NEG_INTERNATIONAL)) { # Do something } When the content of our regex groups file is like this: [6] ^(\+|00)3(0690|0693|0694|0695|0697|0698|0699|36|37|4160|46|47|51160|511691|511693|511696|511891|511893|511896|513|516091|5160921|516093|516096|516391|516392|516393|516396|516591|516592|516593|516596|516691|5166921|516693|516696|5191|5192|5193|5196|538|54388|543 89|546|5475|5477|548|567117|5672|5677|5679|5692|5694|5696|5698|5699|5795|5796|5797|5799|58299|584|5850|5948|5987|5988|5989|59988|59989|620|630|631|670|706|712|725|728|774|776|8591|8592|8595|8597|8598|8599|8620|8621|8630|8631|8640|8641|8643|8649|8650|8651|8660| 8661|8664|8670|8671|869814|906698|93) The above if statment is not matching, although it should match according to any regex validator (for example https://regex101.com/ ) But when I split the above regex in smaller parts, like this: [6] ^(\+|00)306(90|93|94|95|97|98|99) ^(\+|00)336 ^(\+|00)337 ^(\+|00)34160 ^(\+|00)346 ^(\+|00)347 ^(\+|00)351(160|1691|1693|1696|1891|1893|1896|3|6091|60921|6093|6096|6391|6392|6393|6396|6591|6592|6593|6596|6691|66921|6693|6696|91|92|93|96) ^(\+|00)3538 ^(\+|00)354(388|389|6|75|77|8) ^(\+|00)356(7117|72|77|79|92|94|96|98|99) ^(\+|00)357(95|96|97|99) ^(\+|00)358(299|4|50) ^(\+|00)359(48|87|88|89|988|989) ^(\+|00)3620 ^(\+|00)363(0|1) ^(\+|00)3670 ^(\+|00)3706 ^(\+|00)3712 ^(\+|00)372(5|8) ^(\+|00)377(4|6) ^(\+|00)385(91|92|95|97|98|99) ^(\+|00)386(20|21|30|31|40|41|43|49|50|51|60|61|64|70|71|9814) ^(\+|00)3906698 ^(\+|00)393 Then the exact same regex module call as before does result in a match as expected. So to me it looks like the regex module is having trouble with large regex expressions, I did check the logs but there is nothing strange there, that means also the large expression is correctly loaded according to the logs while in fact it is not working. Checked the regex module documentation but there I cannot find any limits regarding to the size of invididual expressions. Is it me missing something or did I discover a bug in de regex module? Best Regards, Tim Balmer Edutel BV

2 2

Re: [SR-Users] Question on the debug change level command with 5.0.x
by Federico Cabiddu 24 Mar '17

24 Mar '17

No, you don't. The debugger module, among other things, allows to set the debug level per module. Regards, Federico On 23 Mar 2017 6:31 pm, "Ginhoux, Patrick" <patrick.ginhoux(a)fr.unisys.com> wrote: Hi, Thanks you so much; it works fine as expected. Now have I to keep the ‘debugger’ module lines in my config ? Cordialement Patrick GINHOUX *De :* sr-users [mailto:sr-users-bounces@lists.sip-router.org] *De la part de* Federico Cabiddu *Envoyé :* jeudi 23 mars 2017 17:38 *À :* Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> *Objet :* Re: [SR-Users] Question on the debug change level command with 5.0.x Hi, you have to load the cfg_rpc module (http://www.kamailio.org/docs/ modules/5.0.x/modules/cfg_rpc.html). Then you can run kamcmd cfg.set_now_int core debug <LEVEL>. Regards, Federico On Thu, Mar 23, 2017 at 5:24 PM, Ginhoux, Patrick < patrick.ginhoux(a)fr.unisys.com> wrote: Hi, Previously, in my kamailio.cfg 3.3.1, the debug level is set using the ‘debug’ option. Then with the MI_FIFO module it was simple to change debug level on the flight using the simple command ‘kamctl fifo debug <new value>’. Moving to kamailio 5.0.x, there is no more MI_FIFO module. In the kamailio.cfg, I still have the ‘debug’ option set; so I’m looking for the command to use to change the debug level. I look at the debugger module that seems the solution. So I add the following lines in y script: debug=4 loadmodule "debugger.so" modparam("debugger", "log_level", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level_mode", 1) Then I try to use the following commands : This one retrieves the debug level set by the debug=4) : [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level core 4 This one changed the debug level for the core module [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level core 2 200 ok Same commands can be used for each of the module loaded: [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level tm 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level tm 2 200 ok [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level dispatcher 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level dispatcher 2 200 ok Each command only applies to a specific module. I don’t find a ‘global’ command that I could use in the same manner as the old ‘kamctl fifo debug’ command. Is there something I missed in my config ? Cordialement Patrick GINHOUX _______________________________________________ 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 _______________________________________________ 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

2 1

Question on the debug change level command with 5.0.x
by Ginhoux, Patrick 23 Mar '17

23 Mar '17

Hi, Previously, in my kamailio.cfg 3.3.1, the debug level is set using the 'debug' option. Then with the MI_FIFO module it was simple to change debug level on the flight using the simple command 'kamctl fifo debug <new value>'. Moving to kamailio 5.0.x, there is no more MI_FIFO module. In the kamailio.cfg, I still have the 'debug' option set; so I'm looking for the command to use to change the debug level. I look at the debugger module that seems the solution. So I add the following lines in y script: debug=4 loadmodule "debugger.so" modparam("debugger", "log_level", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level_mode", 1) Then I try to use the following commands : This one retrieves the debug level set by the debug=4) : [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level core 4 This one changed the debug level for the core module [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level core 2 200 ok Same commands can be used for each of the module loaded: [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level tm 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level tm 2 200 ok [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level dispatcher 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level dispatcher 2 200 ok Each command only applies to a specific module. I don't find a 'global' command that I could use in the same manner as the old 'kamctl fifo debug' command. Is there something I missed in my config ? Cordialement Patrick GINHOUX

2 2

Presence + RLS + XCAP
by Paul Smith 23 Mar '17

23 Mar '17

Longer version of this got blocked by the mail list earlier today… and I have dug a bit deeper since then… I am trying to SUBSCRIBE to a resource-list (rls.so) but the rls_handle_subscribe() is returning with a “list not found" Mar 23 16:36:11 presence kamailio[12536]: DEBUG: rls [subscribe.c:211]: rls_get_service_list(): No rl document found Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 2 columns Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7f70b50ebee8 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[1] at 0x7f70b50afb90 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7f70b50eaf98 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7f70b50cc0a0 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7f70b50cc2d0 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: rls [subscribe.c:634]: rls_handle_subscribe(): list not found - searched for uri <sip:org1a-contacts@**MYDOMAIN**> I have tried various permutations of xml docs in the xcap table for the rls-service and resource-list but nothing have not yet managed to get a success from the rls_handle_subscribe() Latest version of data in my mysql database / xcap table: | 3 | org1a-contacts | **MYDOMAIN** | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@**MYDOMAIN**"/><entry uri="sip:1001@**MYDOMAIN**"><display-name>bob</display-name></entry><entry uri="sip:1003@**MYDOMAIN**"><display-name>test1</display-name></entry><entry uri="sip:1006@**MYDOMAIN**"><display-name>test1006</display-name></entry><entry uri="sip:1007@**MYDOMAIN**"><display-name>test1007</display-name></entry><entry uri="sip:1008@**MYDOMAIN**"><display-name>test1008</display-name></entry><entry uri="sip:1009@**MYDOMAIN**"><display-name>test1009</display-name></entry> <entry uri="sip:1005@**MYDOMAIN**"><display-name>testing1005</display-name></entry> <entry uri="sip:org1a@**MYDOMAIN**"><display-name>testingorg1a</display-name></entry> <entry uri="sip:org1c@**MYDOMAIN**"><display-name>testingorg1c</display-name></entry> <entry uri="sip:dnd1002-6@presence.byphone.co.uk"><display-name>1002 dnd</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10960-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@**MYDOMAIN**/index | 0 | | 4 | org1a-contacts | **MYDOMAIN** | <?xml version="1.0" encoding="UTF-8" standalone="no"?><rls-services xmlns="urn:ietf:params:xml:ns:rls-services"><service uri="sip:org1a-contacts@**MYDOMAIN**"><list name="org1a-contacts"><rl:entry uri="1000_6@**MYDOMAIN**"/><rl:entry uri="org1c@**MYDOMAIN**"/></list><packages><package>presence</package></packages></service></rls-services> | 8 | sr-1490285288-12551-1 | 0 | /xcap-root/rls-services/users/sip:org1a-contacts@**MYDOMAIN**/index | 0 | Is there a reference doc somewhere with an example of a successful SUBSCRIBE and associated XCAP docs for a resource-list? Thanks Paul

1 0

failed to find append_branch
by Anuradha Selvarajan 23 Mar '17

23 Mar '17

Hello, I am a new user to kamailio. I am trying to use the fork.cfg to send sip registration to 2 destination. I get the following error but I have added loadmodule tm.so, why am I still getting this error? Mar 22 15:53:48 emps116 kamailio[14065]: ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command append_branch (params 1) Mar 22 15:53:48 emps116 kamailio[14065]: : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/fork.cfg, line 45, column 43: unknown command, mi Mar 22 15:53:48 emps116 kamailio[14065]: ERROR: bad config file (1 errors) Mar 22 15:53:48 emps116 kamailio[14065]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules Mar 22 15:53:48 emps116 kamailio[14065]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Mar 22 15:53:48 emps116 kamailio[14065]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 22 15:53:48 emps116 systemd[1]: kamailio.service: Control process exited, code=exited status=255 Mar 22 15:53:48 emps116 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. My script looks like this (default one): mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules" loadmodule "sl.so" loadmodule "tm.so" . . . . route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); exit; }; # try these two destinations first in parallel; the second # destination is targeted to sink port -- that will make ser # wait until timer hits seturi("sip:nobody@kamailio.org"); append_branch("sip:parallel@kamailio.org"); # if we do not get a positive reply, continue at reply_route[1] #t_on_failure("1"); # forward the request to all destinations in destination set now t_relay(); } Appreciate any help! Thanks Anu

3 3

Presence + RLS + XCAP
by Paul Smith 23 Mar '17

23 Mar '17

Hi I am struggling to figure out how to build and test an integrated presence server with Resource-List (rls.so) and XCAP (xcap_server.so). I am confused about what is meant to happen and how to debug it. Not sure if I am suffering from incompatibilities, configuration errors, or bugs… I have not found an up to date guide on configuring an integrated presence, rls, xcap server so I have used old tutorials, module documentation and some course code review … but I’m pretty confused now! I am using Snom phones and Jitsi (on Mac) to test. I have a kamailio Registrar which proxies all presence messages to the separate kamailio 4.2 Presence server. REQUIREMENT : Contact Presence ============================= I would like to get a SUBSCRIBE from the Snom phone to subscribe to a contact-list resource list, and to receive a NOTIFY with all of the presence states for those contacts. Relevant snom settings? Contact List URI: this generates a SUBSCRIBE with event:presence I believe that: ========== presence module should be managing the individual user presence and dialog states xcap module should allow me to use HTTP / curl to edit the resource lists rls module should be managing the mapping from a resource list subscription to the individual presence states and generating the aggregated NOTIFY messages. each device should then be able to SUBSCRIBE to a resource list uri and then receive an aggregated NOTIFY for all of the watched entities. Current status: =========== SUBSCRIBE sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk> SIP/2.0 v: SIP/2.0/UDP 192.168.1.15:59594;branch=z9hG4bK-qllzlgs3xf43;rport f: <sip:org1a@*****.co.uk <sip:org1a@*****.co.uk>>;tag=wwasqc6lmb t: <sip:org1a-contacts@****.co.uk <sip:org1a-contacts@****.co.uk>> i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 1 SUBSCRIBE Max-Forwards: 70 User-Agent: snomD375/8.9.3.46 m: <sip:org1a@192.168.1.15:59594 <sip:org1a@192.168.1.15:59594>>;reg-id=1 Event: presence Accept: application/pidf+xml, application/rlmi+xml, multipart/related Supported: eventlist Expires: 3600 l: 0 … auth handshake SIP/2.0 202 OK Record-Route: <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes>> Record-Route: <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes>> v: SIP/2.0/UDP 192.168.1.15:59594;received=**recvip**;branch=z9hG4bK-8wufiv1ua3ne;rport=60554 f: <sip:org1a@****.co.uk <sip:org1a@****.co.uk>>;tag=wwasqc6lmb t: <sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk>>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d30e i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 2 SUBSCRIBE Expires: 3600 Contact: <sip:10.210.160.16:5060;transport=udp <sip:10.210.160.16:5060;transport=udp>> Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 … and then the snom receives an empty NOTIFY. NOTIFY sip:org1a@192.168.1.15:58325 <sip:org1a@192.168.1.15:58325> SIP/2.0 Via: SIP/2.0/UDP 109.104.108.18;branch=z9hG4bKc54a.20db45097fb44595ccd5a30827ca2a62.0 Via: SIP/2.0/UDP 10.210.160.16;rport=5060;branch=z9hG4bKc54a.40150886000000000000000000000000.0 To: <sip:org1a@****.co.uk <sip:org1a@****.co.uk>>;tag=sx4ne9jpcl From: <sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk>>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-ed89 CSeq: 2 NOTIFY Call-ID: 313439303237323938323235353133-2du221e6okvf Content-Length: 0 User-Agent: kamailio (4.2.0 (x86_64/linux)) Max-Forwards: 69 Event: presence Contact: <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1 <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1>> Subscription-State: active;expires=3600 The Subscription is showing up in the watchers table, not the rls_watchers table, so I believe it is being rejected by the rls_handle_subscribe() and then parsed by the presence module. Is that correct? There is noting in rls_watchers or rls_presentity tables. Should there be? I have a rows in the XCAP table for the Resource-List curl -d @/tmp/contacts-resource-list.xml -vvv -X PUT http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact… <http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact…> relevant row in xcap table: | 3 | org1a-contacts | *****.co.uk <http://co.uk/> | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@****.co.uk <http://co.uk/>"/><entry uri="sip:1001@****.co.uk <http://co.uk/>"><display-name>bob</display-name></entry><entry uri="sip:1003@****.co.uk <http://co.uk/>"><display-name>test1</display-name></entry><entry uri="sip:1006@****co.uk <http://co.uk/>"><display-name>test1006</display-name></entry><entry uri="sip:1007@****co.uk <http://co.uk/>"><display-name>test1007</display-name></entry><entry uri="sip:1008@h****.co.uk <http://co.uk/>"><display-name>test1008</display-name></entry><entry uri="sip:1009@****.co.uk <http://co.uk/>"><display-name>test1009</display-name></entry> <entry uri="sip:1005@****.co.uk <http://co.uk/>"><display-name>testing1005</display-name></entry> <entry uri=“sip:org1a@****.co.uk <http://co.uk/>"><display-name>testingorg1a</display-name></entry> <entry uri=“sip:org1c@****.co.uk <http://co.uk/>"><display-name>testingorg1c</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10965-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@****co.uk/index <http://co.uk/index> | 0 | I have rows in the presence table for the entities that I want to track e.g. | | | 2889 | org1c | ****.co.uk <http://co.uk/> | presence | a.1490270411.10942.42.1083 | 1490277099 | 1490273499 | <?xml version="1.0 " encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:im="urn:ietf:params:xml:ns:pidf:im" entity=“pres:org1c@*****.co.uk <http://co.uk/>"> <tuple id="snom370-000413260935"> <status><basic>open</basic> <im:im <im:im>>Available</im:im> </status> <contact priority="1.00">sip:org1c@****.co.uk <http://co.uk/></contact> <note xml:lang="en">Available</note> </tuple></presence> I am running kamaiio 4.2 Question : Should I have an resource-services row too? Or does rls.so infer the relationship using the URI? When I use jitsi to manipulate its contact list it only inserts 2 rows into the xcap table and I cannot get a SIP Subscribe to connect to that resource list either. Question : What should the curl PUT commands and XML docs look like to build a valid RLS service (is there a good example anywhere)? Question : Is there an example config or doc anywhere showing how to build integrated presence, rls, xcap? Question : Has anyone got this working with Snom and or other handsets… the standards and implementations seem vague / inconsistent? Note: There is another Snom setting (Extension Monitoring Call Pickup List URI:) which produces a SUBSCRIBE with Event:dialog but which should allow a button/lamp/callpickup/BLF for the buddy contact. I’d like to get that working too, but recognise that rls.so does not handle dialog messages by default, so I’m starting with Presence... Paul Paul Smith 1 Lanyon Quay, Belfast, BT1 3LG UK: 02890270050 RoI: 1800855111 paul.smith(a)claritytele.com <mailto:paul.smith@claritytele.com%0b> www.claritytele.com <http://claritytele.com/> This email and any attachment, is confidential. If you have received this in error, please delete it from your system, do not use or disclose the information in any way, and notify me immediately. The contents of this message may contain personal views which are not the views of Clarity Telecom, unless specifically stated. > >> >>

1 0

Status of db_cassandra
by Markus Bönke 23 Mar '17

23 Mar '17

Hello, We are thinking about to use kamailio as a sip registrar with cassandra as db backend. As I can see in the documentation for kamailio 5.0 it is only tested with Casandra 1.1.6 and 1.0.1, current version is 3.10 in the meantime. Is anyone using this module with newer versions of Cassandra? Is it stable? Thanks and regards Markus

4 6

looks like dp_translate with pvs in repl_exp is badly broken
by Juha Heinanen 23 Mar '17

23 Mar '17

when match_exp of dp 100 is ^#test and repl_exp is sip:$rU@$rd this code: xlog("L_INFO", "****** ru = $ru\n"); dp_translate(100, "$rU/$var(res)"); xlog("L_INFO", "****** res = $var(res)\n"); produces: Mar 22 08:21:17 lohi /usr/bin/sip-proxy[7543]: INFO: ****** ru = sip:#test@test.tutpro.com Mar 22 08:21:17 lohi /usr/bin/sip-proxy[7543]: INFO: ****** res = sip:$rU@$rd -- juha

4 6

getting source IP from Contact header
by Ali Taher 22 Mar '17

22 Mar '17

Hello everyone, I'm trying to get the host part from the contact header in the sip invite packet. Following is the contact header format: Contact: <sip:user_name@192.168.1.1:5060> What I want as result is 192.168.1.1 I searched among the pseudovairables and found $ct variable which reference to the body of contact header. (<sip:user_name@192.168.1.1:5060>) Is there a way to get directly the host (192.168.1.1) without applying transformation at the level of $ct. If not what transformation can achieve it. Thanks in advance. Regards, Ali

3 2

Alternative of SIPp for WebRTC2SIP load-test
by Jade SZ 21 Mar '17

21 Mar '17

Hi Guys, SIPp is great in testing pure SIP udp/tcp/tls but I was wondering if there is a similar tool for stress testing Kamailio websocket2sip. I am trying to load-test webrtc2sip kamailio's implementation closest to the real scenario. One way would be to fork multiple process of webrtc2sip client in loop. But would really like to hear how are kamailio webrtc users/community load-testing it for benchmarking their servers. Also roughly how much CPS they have achieved using single kamailio with webrtc enabled (share server specs as well) Thanks in advance. Regards, JSZ

2 1

do we need high available IP for S-CSCF?
by Valentin Christoph 21 Mar '17

21 Mar '17

Hi all, Maybe I need a small hint. I have got a configuration of 2 S-CSCF instances (both kamailio 4.4.5 with S-CSCF configuration, each on a separate virtual machine) in load sharing mode (50:50). Both instances work on the same "scscf" data base, so that it is only one S-CSCF from a logical point of view. Now I get into troubles with registration: * P-CSCF sends REGISTER to first S-CSCF * First S-CSCF answers with 401 Unauthorized an challenges the UE. So far everything OK. * Second REGISTER is sent to second S-CSCF (DNS SRV Load Balancing). * Second S-CSCF answers again with 401 Unauthorized (Error) * This goes back and forth until UE gives up to register This problem exists sporadically (approx. every fifth registration). Now the question: is it worth to debug this configuration or should we rather switch to high available IP configuration (if we want to use kamailio)? What's your opinion/experiences? Thanks Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

Siremis 4.3.0 with Apache 2.4
by kai.ohnacker＠cbc.de 21 Mar '17

21 Mar '17

Hello Community, I installed the Kamailio 4.4 and try to install the Siremis GUI based on apache 2.4. If I try to start the Siremis Wizard, the Webserver display the following line: "location.href='$url'"; exit; } ?>". The starting page of apache 2.4 is working fine, but the /siremis page doesn't. Due the installation I take the dedicated apache 2.4 commands. Are there some further requirements? Some ideas? OS: Ubuntu 16.04 Thanks a lot. Cheers, Kai

2 2

No cdr after Kamailio 5.0 restart
by Kordován Szabolcs 21 Mar '17

21 Mar '17

Hi, I use acc module for create cdr into db and also make acc log into db. After restart Kamailio cdr won't be created but acc log will be inserted into db. In the syslog also I see just the acc log. The dialog will be remove from db. I have read many threads. I know that it should work. Thanks for help! Regards, Szabolcs >From my configs: #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 modparam("acc", "db_url", DBURL) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "log_level", 3) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "acc_prepare_always", 1) modparam("acc", "time_mode", 1) modparam("acc", "time_attr", "seconds") modparam("acc", "time_exten", "microsecs") modparam("acc", "reason_from_hf", 1) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdr_extra", "callid=$ci;src_user=$dlg_var(fU);src_domain=$dlg_var(fd);dst_ouser=$dlg_var(tU);dst_user=$dlg_var(rU);dst_domain=$dlg_var(rd);src_ip=$dlg_var(si);peer_in=$dlg_var(peerin);peer_out=$dlg_var(peerout)") request_route { <------> <------>$var(loose) = loose_route(); <------>$var(check_trans) = t_check_trans(); <------>sip_trace(); <------>setflag(FLT_SIPTRACE); <------>dlg_manage(); <------># handle retransmissions <------>if(t_precheck_trans()) { <------> xlogl("LOG_LOCAL0", "L_ALERT", "================== kamaty $rm route $ci ==== $rm ==== precheck_trans exit\n"); <------> exit; <------>} <------>if(is_method("OPTIONS")) {. <------> # send reply for each options request. <------> sl_send_reply("200", "ok");. <------> exit();. <------>} <------># CANCEL processing <------>if (is_method("CANCEL")) { <------><------>if (t_check_trans()) { <------><------> route(RELAY); <------><------>} <------><------>exit; <------>} <------>if ( is_method("ACK") ) { <------><------>if ( $var(check_trans) ) { <------><------> route(RELAY); <------><------>} <------><------>exit; <------>} <------>if (is_method("BYE")) { <------> if ($var(loose)) { <------><------>setflag(FLT_ACC); # do accounting ... <------><------>setflag(FLT_ACCFAILED); # ... even if the transaction fails <------><------>setflag(FLT_ACCMISSED); <------><------>route(RELAY); <------> <-->exit; <------> } <------> exit; <------>} etc.

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Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 21 Mar '17

21 Mar '17

Hi, There was a mistake in the file attached ; the short kamailio.cfg file (that makes kamailio to start) was not the correct one; attached the correct one. Have anyone an idea of this problem ? Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : lundi 20 mars 2017 17:58 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : RE: ht_db_load_table(): key type must be string (type=6) Hi, To figure out what causes this issue, I work to shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>) <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX

1 0

Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 20 Mar '17

20 Mar '17

Hi, To figure out what causes this issue, I work to shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX

1 0

ACC - Cdr enabled
by Diego Nadares 20 Mar '17

20 Mar '17

Hi guys, I have configured to use cdr with extra fields modparam("acc", "cdr_extra", * "sip_code=$rs;sip_reason=$rr;"* "hangup_src=$dlg_var(hangup_disposition);" "prefix=$dlg_var(prefix);gos=$dlg_var(gos);endpoint_type=$ dlg_var(endpoint_type);" "ip_a=$dlg_var(ip_a);ip_b=$dlg_var(ip_b);" Is this the right way to always save the hangup cause? Or is another better way? Because, when call is finished with a bye both fields are empty. The other thing is that I need to save, start_time and end_time with seconds.microseconds (1490013250.601707). Is this possible? Thanks! Diego

2 5

Is File Sharing between two Extensions Possible?
by Shahid Mehmood 20 Mar '17

20 Mar '17

Dear all, I want to share (any kind of) files between two extensions. Is it possible? If it is, then how ??? please elaborate with example. -- Shahid Mehmood

1 0

cisco spa502g issues
by ben＠wtf.com.au 19 Mar '17

19 Mar '17

Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@<our domain> Contact:: sip:5546@<OUR OFFICE IP>:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018(a)10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:<OUR SR IP>:5060 Methods:: 6815 AOR:: 5590(a)sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db(a)10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip:<OUR OFFICE IP>:1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp:<OUR SR IP>:5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben.

3 4

Generate extra cdr per dialog
by Diego Nadares 18 Mar '17

18 Mar '17

Hi Guys, I'm using acc module for accounting with the following conf: .... modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_facility", "LOG_LOCAL1") modparam("acc", "cdr_end_id", "end_time_dt") modparam("acc", "cdr_extra", "sip_code=$rs;sip_reason=$rr;" "hangup_src=$dlg_var(hangup_disposition);" "prefix=$dlg_var(prefix);gos=$dlg_var(gos);endpoint_type=$dlg_var(endpoint_type);" "ip_a=$dlg_var(ip_a);ip_b=$dlg_var(ip_b);" .... It works fine. At the end of the call the cdr is generated ok. BUT In MANAGE_FAILURE, If the first call failed and if there are more dests, we are generating a new call in the same dialog (Some of the dialog vars like ip_a, ip_b can change for the new dest ) I would like to generate both cdrs, the one that failed and the new one (with some dlg vars with new data). if(rtjson_next_route() && t_check_status("503")) { $avp(current_route) = $avp(current_route)+1; t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit; } Right now just the last call cdr is saved. I all ready made this work with db_acc (transaction oriented) but I think in my case it will be more reliable with cdr (dialog oriented). Thanks in advance! Diego.

1 1

f_malloc WARNINGS in syslog
by Diego Nadares 17 Mar '17

17 Mar '17

Hi Guys, I'm seeing a lot of warnings in my syslog. Is this normal? Or maybe I am doing something wrong in my cfg? .... Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:444]: fm_malloc(): fm_malloc(0x7f68e5199010, 8) called from nathelper: nathelper.c: nh_timer(2078) Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:527]: fm_malloc(): fm_malloc(0x7f68e5199010, 8) returns address 0x7f68e52a2d50 Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:588]: fm_free(): fm_free(0x7f68e5199010, 0x7f68e52a2d50), called from nathelper: nathelper.c: nh_timer(2222) Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:607]: fm_free(): fm_free: freeing block alloc'ed from nathelper: nathelper.c: nh_timer(2078) ..... Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:607]: fm_free(): fm_free: freeing block alloc'ed from core: xavp.c: xavp_new_value(94) Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:588]: fm_free(): fm_free(0x7f68d9d51000, 0x7f68da069ca8), called from tm: h_table.c: free_cell_helper(246) Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:607]: fm_free(): fm_free: freeing block alloc'ed from tm: h_table.c: build_cell(317) Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:588]: fm_free(): fm_free(0x7f68d9d51000, 0x7f68da031f00), called from tm: h_table.c: free_cell_helper(150) .... Thanks in advance. Diego

2 2

ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX

1 0

Error on <ds_probing_threshhold> of type <2> not found in module <dispatcher> and SCTP API Not initialized
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, I'm migrating my kamailio.cfg from the 3.3.1 release to 5.0.0 and encounter the following errors : Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: Not starting : invalid configuration file! Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <ds_probing_threshhold> of type <2> not found in module <dispatcher> Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 295, column 49: Can't set module parameter Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: ERROR: bad config file (1 errors) Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. Below the extract of my kamailio.cfg file : modparam("dispatcher", "list_file", DISPATCHER_FILE) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold",3) modparam("dispatcher", "flags", 2) modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "ds_hash_size", 8) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "force_dst", 1) For the dispatcher error : I don't find any change (eg in the Changelog for 5.0.0.) regarding the dispatcher module. So is it a bug in the dispatcher module or have I to change it to something else ? For the SCTP API error : I don't know what to do here. Thanks to people that can help me on these 2 errors Cordialement Patrick GINHOUX

2 3

Configuration to fetch SIP response body.
by vivek＠advaitamtech.com 17 Mar '17

17 Mar '17

Hi, Is there any configuration to fetch SIP response body received?. My requirement is to fetch the SIP response body and save it into database. Thanks, Vivek

3 3

Search for kamailio PERL module for 5.0 level
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, I'm a beginner in the use of kamailio product, so I try below to explain my problem with my words: 1) PERL module issue : Some years ago a SIP Router proxy running RedHat 6.2 was developped using the kamailio version 3.3 and the following packages used : kamailio-3.3.2-1.1.1.x86_64.rpm kamailio-snmpstats-3.3.2-1.1.1.x86_64.rpm kamailio-mysql-3.3.2-1.1.1.x86_64.rpm kamailio-perl-3.3.2-1.1.1.x86_64.rpm We have the project to migrate this SIP Proxy to RedHat 7.1 and used the latest Kamailio v5.0.0 Release: kamailio-5.0.0-24.1.x86_64.rpm kamailio-mysql-5.0.0-24.1.x86_64.rpm kamailio-snmpstats-5.0.0-24.1.x86_64.rpm However I don't find the kamalio-perl package for RedHat 7.1. I see that the perl module has been renamed to 'app_perl' but I don't find it either for kamailio 5.0.0. 2) Kamailio.cfg script Our proxy used the kamailio.cfg from the sample of the 3.3.3 release. So can I reuse the same 3.3.3 kamailio.cfg file with the 5.0.0 release or should I migrate (rewrite it) from the 5.0.0 sample ? So I'm looking for some help on these issues. Is there people that had this case ? Thanks in advance for any answer. Cordialement Patrick GINHOUX PS : I just create my account today but don't get the email to validate it.

2 6

(no subject)
by Alberto Llamas 17 Mar '17

17 Mar '17

Hi Guys, Who knows and can recommend a company a company where I could order Toll free numbers for EU?. I really appreciate any hint and sorry for the noise. Regards, -- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA *"Internet is all about share"*

2 1

cisco spa502g issues
by Ben Kramer 17 Mar '17

17 Mar '17

Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@<our domain> Contact:: sip:5546@<OUR OFFICE IP>:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018(a)10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:<OUR SR IP>:5060 Methods:: 6815 AOR:: 5590(a)sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db(a)10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip:<OUR OFFICE IP>:1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp:<OUR SR IP>:5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben.

1 0

send_all blocks indefinitely
by Robert 16 Mar '17

16 Mar '17

H all, Recently, when I'm trying to load test Kamailio with a single proxy, it hung at around 15000 TPS. I attached gdb to one of the kamailio processes and I saw this: (gdb) where #0 0x00007fa8872366ad in __libc_send (fd=13, buf=0x7ffe6e0e5350, n=16, flags=-2027723091) at ../sysdeps/unix/sysv/linux/x86_64/send.c:27 #1 0x00000000004eb635 in send_all (socket=13, data=0x7ffe6e0e5350, data_len=16) at pass_fd.c:133 #2 0x00007fa4155a1b69 in tcp_close_connection (conn_id=28022) at tcp_funcs.c:199 #3 0x00007fa4155a363f in cleanup (start=0x7fa4540c3ab8, end=0x7fa42f92dca0) at tcp_funcs.c:337 #4 0x00007fa4155a1496 in timer_clean_tcp_connections (ticks=18583555, param=0x0) at tcp_funcs.c:165 #5 0x0000000000617f79 in fork_basic_timer (child_id=-1, desc=0x7fa4155a88e0 "Connections cleanup timer", make_sock=1, f=0x7fa4155a0ac6 <timer_clean_tcp_connections>, param=0x0, interval=30) at timer_proc.c:79 #6 0x00007fa4155a5816 in child_init (rank=0) at tcp_mod.c:176 tcp_func.c is something I wrote myself, which is basically a timer that runs periodically to try to clean up tcp connections that I no longer need. The code for tcp_close_connection is taken directly from usrloc > urecord.c > close_connection() >From the gdb output above, it seems like send_all is blocked and never returned, which led to my proxy to hang. In kamailio/src/core/pass_fd.c, I saw this: ... /** sends all data (takes care of signals) (assumes blocking fd) * returns number of bytes sent or < 0 for an error */ int send_all(int socket, void* data, int data_len) But in kamailio/doc/misc/TODO, I saw this: ... x [tcp] make send_all, send non-blocking ready ? x [tcp] tcp_send w/ timeout (to avoid waiting forever to send DOS) So, my question is, is send_all() suppose to block or not? I'm fairly new to how tcp and socket connection works. Any help will be appreciated! Thanks, Robert -- View this message in context: http://sip-router.1086192.n5.nabble.com/send-all-blocks-indefinitely-tp1569… Sent from the Users mailing list archive at Nabble.com.

1 0

Cdr on shutdown
by Diego Nadares 16 Mar '17

16 Mar '17

Hi Guys, Is it possible to generate cdrs or to save dialogs to db when kamailio receives a kill signal from terminal? Thanks in advance. Diego

3 5

siptrace duplicate both incoming and outgoing messages
by Grant Bagdasarian 16 Mar '17

16 Mar '17

Hi, Is it possible to instruct Kamailio to duplicate both incoming and outgoing messages automatically? If not, if I set the flag and explicitly call sip_trace, will both messages be duplicated to the capture server? Regards, Grant Bagdasarian CM

2 1

mi_fifo: No module matching <jsonrpcs> found
by Mostafa Zare 15 Mar '17

15 Mar '17

hi guys i'm newbie to kamailio. i've installed kamailio version 5 (and all the modules) from pkgs on a jessie debian. i have the below config on my kamailio.cfg when i try to start kamailio service it throws the following error. what's the problem? - systemctl status kamailio: kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled) Active: failed (Result: exit-code) since Wed 2017-03-15 19:01:17 IRST; 10s ago Process: 9377 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) Main PID: 1576 (code=exited, status=0/SUCCESS) Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <jsonrpcs> found Mar 15 19:01:17 kamailio-debian kamailio[9377]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 103, column 72: Can't set module parameter Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <ds_probing_threshhold> of type <2> not found in module <dispatcher> Mar 15 19:01:17 kamailio-debian kamailio[9377]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 135, column 50: Can't set module parameter Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: bad config file (4 errors) Mar 15 19:01:17 kamailio-debian kamailio[9377]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 15 19:01:17 kamailio-debian kamailio[9377]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Mar 15 19:01:17 kamailio-debian systemd[1]: kamailio.service: control process exited, code=exited status=255 Mar 15 19:01:17 kamailio-debian systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Mar 15 19:01:17 kamailio-debian systemd[1]: Unit kamailio.service entered failed state. - kamailio.cfg: # *** To run in debug mode: # - define WITH_DEBUG # #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 # log_facility=LOG_LOCAL0 # Ds increasing debug... debug=2 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:0.0.0.0:5060 # listen=127.0.0.1:5060 sip_warning=no ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- # modparam("dispatcher", "db_url", DBURL) # modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # dispatch destinations route(DISPATCH); # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; sl_send_reply("404", "No registrar"); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # Destination types: # http://kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatcher.f… # 4 = round robin # 9 = weight-based # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "9")) { send_reply("404", "No destination"); exit; } t_set_fr(0,2000); t_on_failure("ROUTE_ADVANCE"); route(RELAY); exit; } event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } event_route[dispatcher:dst-up] { xlog("L_ERR", "Destination up: $rm $ru\n"); } failure_route[ROUTE_ADVANCE] { if (t_is_canceled()) { exit; } xlog("L_NOTICE", "Media server $du failed to answer, selecting other one!\n"); if (t_check_status("503") || (t_branch_timeout() and !t_branch_replied())) { ds_mark_dst("ip"); # select the new destination if(ds_next_dst()) { # again set local timeout for reply # xlog("L_NOTICE", "Selecting next route upon failure\n"); t_set_fr(0,2000); t_on_failure("ROUTE_ADVANCE"); route(RELAY); exit; } else { #last available node failed to reply, no other destinations available send_reply("404", "No destination"); xlog("L_NOTICE","No destination, sent 404\n"); exit; } } else { xlog("L_NOTICE","Unhandled failured route!\n"); } }

2 1

SIP contact header question
by Jack Davis 15 Mar '17

15 Mar '17

Greetings, I have a general question about the usage of SIP contact headers in the context of using Kamailio as a SIP proxy. ________________________________ [ A ] --> [ Kamailio B ] ---> [ C ] Node A originates a SIP invite, containing a valid via header and URI while setting the contact address to a user@itself and delivers it to Kamailio B which is acting as a SIP proxy. Kamailio B then uses dispatcher routing to direct the Invite to node C, adding a via line with its own information as well as a record-route header with its own proxy information but retaining the same contact address from A. Node C establishes the call and then sends a re-invite to the Kamailio B proxy which is in turn sent to Node A. Node A responds with a 200 OK The problem arises when Node C tries to send an Ack in response to this 200 OK. The ack is being sent to the Contact address, rather than the routing already established in the initial dialog. ________________________________ My question is: should kamailio be rewriting this contact address with its own? Is that the best practice? My understanding is that the contact header is more so related to future requests within the same dialog ONLY when a record-route is not used. I would appreciate any clarification on the RFC or best practices in this scenario. Thank you, Jack Davis

5 12

Kamailio does'nt respond to packets from specific addresses
by עדי ויזגן 15 Mar '17

15 Mar '17

Hi everyone, I have the following problem: I have a Kamailio server that is connected to several Asterisk servers for load balacing in a LAN network with the IP's of 192.168.16.X. I recently connected with another company that is supposed to send me calls through the Kamailio server. The problem is this: I am connected to then with a cable from their network to mine. My port has the address 10.9.18.198 and they come from the address 172.18.177.18. When I get packets from their address (172...) I can see them getting to my Kamailio server, but the Kamailio does not respond. It only happens when I get packets from 172 source address. If I get from 192 or 10 adresses from my LAN network it works, and if I get packets from real world IP's through my WAN it works. Does anyone have any idea why the Kamailio does'nt respond to packets from 172 adresses? And how to fix this? Thanks in advance!

2 3

Kamailio World 2017 - A selection of presentations
by Daniel-Constantin Mierla 15 Mar '17

15 Mar '17

Hello, I would like to announce that many of the presentations at Kamailio World 2017 (May 8-10, in Berlin, Germany) are now listed at: - https://www.kamailioworld.com/k05/schedule/ Just several are not published yet, working to get all their details in place and expect to have the first draft of the schedule very soon. This edition takes place again at the same location in the famous city center of Berlin, the number of seats is limited, last year the event was fully booked. Do not delay too much your registration if you want to attend this year: - https://www.kamailioworld.com/k05/registration/ Thanks to the sponsors (https://www.kamailioworld.com/k05/sponsors/), it looks like we will be able to offer again some grants for students or underrepresented groups. More details will be sent out in the near future. Looking forward to meeting many of you in Berlin! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # RTPProxy control 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; if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel

1 0

Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # RTPProxy control 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; if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel

1 0

rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control 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"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

2 1

Is kamailio support load balancing for websockets
by SureshTummala 14 Mar '17

14 Mar '17

Hi, I want to use kamailio as a load balancer for web sockets. Is kamailio supports LB functionality for web-sockets? I can able to use kamailio as a proxy and can make calls with websockets. Now i want to add LB functionality. Setup: UA1 -->LB -->Proxy UA2 <--LB <--Proxy LB is not forwarding the packets from Proxy to UA Please help me Regards,Suresh Tummala

1 0

Several Asterisk on this same IP
by przeqpiciel 14 Mar '17

14 Mar '17

I would like to ask you how you deal with several asterisks and kamailio on that same IP address, I have installation where i route 5060 to internal server with kamailio and I would like to route RTP to asterisks, but is any way to get around a problem with RTP ports collisions?

6 19

how to unregister an remote registration at uac module
by José Lopes 13 Mar '17

13 Mar '17

Hello, I am using the module uac to do remote registration. But sometimes i need to do unregister, to control or debug the registration process. I tried using kamcmd with uac.reg_disable, but this doesn't unregister. There is any way to do it? Best regards, Jose Lopes

2 2

Segmentation fault - dialog RPC
by pars3c 13 Mar '17

13 Mar '17

Dear all, many times, when i call dialog_list on dialog module with jsonrpc, kamailio crash. The version i use is: 4.4.2 Thanks for any suggestion. #0 0x00007f2a706eb00a in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 1666 if ((!value || (STR_EQ(*value, ph->value))) && ph->dlg) { The dump , with bt full is this : #0 0x00007f2a706eb00a in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 profile = 0x7f2a5098fb40 ph = 0x0 i = 10 #1 0x00007f2a706eb6e4 in rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940) at dialog.c:1772 profile_name = {s = 0x1c78b00 "outbound", len = 8} value = {s = 0x0, len = 0} #2 0x00007f2a4fbe45c6 in jsonrpc_dispatch (msg=0x7fffb9fa9f10, s1=0x0, s2=0x0) at jsonrpc-s_mod.c:1091 rpce = 0x7f2a75609360 ctx = 0x7f2a4fdf9940 ret = 0 nj = 0x1c76af0 val = {s = 0x1c77b20 "dlg.profile_list", len = 16} __FUNCTION__ = "jsonrpc_dispatch" #3 0x000000000041f4ad in do_action (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1054 ret = -5 v = 4288256 dst = {send_sock = 0x1b9fa9b23, to = {s = {sa_family = 14080, sa_data = "Xu*\177\000\000\b7Xu*\177\000"}, sin = { sin_family = 14080, sin_port = 30040, sin_addr = {s_addr = 32554}, sin_zero = "\b7Xu*\177\000"}, sin6 = { sin6_family = 14080, sin6_port = 30040, sin6_flowinfo = 32554, sin6_addr = {__in6_u = { __u6_addr8 = "\b7Xu*\177\000\000S \205\000\060\000\000", __u6_addr16 = {14088, 30040, 32554, 0, 8275, 133, 48, 0}, __u6_addr32 = {1968715528, 32554, 8724563, 48}}}, sin6_scope_id = 26}}, id = 0, proto = 16 '\020', send_flags = { f = 155 '\233', blst_imask = 250 '\372'}} tmp = 0x7fffb9fab700 "\t" new_uri = 0x0 end = 0x0 crt = 0x7fffb9fa92d0 "" cmd = 0x7f2a754adcc0 len = 0 user = 0 uri = {user = {s = 0x12c <Address 0x12c out of bounds>, len = 0}, passwd = {s = 0x7fffb9fa92d0 "", len = 1969341184}, host = { s = 0x7f2a75475010 "\001", len = 4742114}, port = {s = 0x0, len = 0}, params = { s = 0x98206f7473 <Address 0x98206f7473 out of bounds>, len = 1969341256}, sip_params = { s = 0x7fffb9fa91e0 "`\222\372\271\377\177", len = 6538286}, headers = {s = 0x1c2b6f0 "\300\300\302\001", len = 0}, port_no = 50838, proto = 30049, type = 32554, flags = 0, transport = {s = 0x0, len = -940795965}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x6222286e00000000 <Address 0x6222286e00000000 out of bounds>, len = 577270895}, maddr = { s = 0x3a746e65696c6309 <Address 0x3a746e65696c6309 out of bounds>, len = 0}, method = {s = 0x75c5f0 "core", len = -1174760864}, lr = {s = 0x764661 "rval_destroy", len = 7718524}, r2 = {s = 0x7f2a7561c348 "\002", len = 1967607824}, gr = {s = 0x7fffb9fa91b8 "H\303au*\177", len = 1967607824}, transport_val = {s = 0x7f2a7561c300 "\300\004", len = 1969341256}, ttl_val = {s = 0x7f2a7557e200 "\020", len = -1174760316}, user_param_val = {s = 0x7f2a75583700 "\002", len = 1969341256}, maddr_val = {s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, method_val = { ---Type <return> to continue, or q <return> to quit--- s = 0x0, len = 0}, lr_val = {s = 0x7fffb9fa9260 "Ð\372\271\377\177", len = 5581478}, r2_val = {s = 0x0, len = 0}, gr_val = { s = 0x3a28d <Address 0x3a28d out of bounds>, len = 1969177728}} next_hop = {user = {s = 0x1c2b6f0 "\300\300\302\001", len = 1969330344}, passwd = { s = 0x69692c22 <Address 0x69692c22 out of bounds>, len = 29537760}, host = { s = 0x400000000 <Address 0x400000000 out of bounds>, len = -1174761304}, port = { s = 0x71657228 <Address 0x71657228 out of bounds>, len = 1969342102}, params = {s = 0x0, len = 1}, sip_params = {s = 0x0, len = -1174761328}, headers = {s = 0x1 <Address 0x1 out of bounds>, len = 2}, port_no = 40672, proto = 47610, type = 32767, flags = 3120206016, transport = {s = 0x7a6563697220616c <Address 0x7a6563697220616c out of bounds>, len = 1701736297}, ttl = { s = 0x7f2a75580848 "\001", len = -1174760828}, user_param = {s = 0x7fffb9fa9f10 "\b\005", len = -1174757840}, maddr = { s = 0x65722e2e223a7069 <Address 0x65722e2e223a7069 out of bounds>, len = 1702129517}, method = {s = 0x7f2a7561c696 "/sip", len = 29855024}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x500000004 <Address 0x500000004 out of bounds>, len = 4}, transport_val = {s = 0x400000004 <Address 0x400000004 out of bounds>, len = 6552774}, ttl_val = { s = 0x7fffb9fa9060 "", len = 1967607824}, user_param_val = {s = 0x1000001 <Address 0x1000001 out of bounds>, len = 1967607824}, maddr_val = {s = 0x7fffb9fa90b0 "\020PGu*\177", len = 1956545302}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14985000}, r2_val = { s = 0x165e80 <Address 0x165e80 out of bounds>, len = 1792216}, gr_val = {s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}} u = 0x7f2a75583f30 port = 0 dst_host = 0x7f2a75583700 i = 32767 flags = 0 avp = 0x7fffb9fa9e30 st = {flags = 151652617, id = 29193, name = {n = 0, s = {s = 0x6e65090a00000000 <Address 0x6e65090a00000000 out of bounds>, len = -1174761256}, re = 0x6e65090a00000000}, avp = 0x7fffb9fa9000} sct = 0x7fffb9fa94d0 sjt = 0x7fffb9fa9f10 rve = 0x416f00 mct = 0x7fffb9fab700 rv = 0x92922726f727265 rv1 = 0x0 c1 = {cache_type = 3120205776, val_type = 32767, c = {avp_val = {n = -940803561, s = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, re = 0x31c7ec7a17}, pval = {rs = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, ri = -1174757840, flags = 32767}}, i2s = "vp(s:inserver_erPor)\","} s = {s = 0x1c2b6f0 "\300\300\302\001", len = 0} srevp = {0x0, 0x0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, ---Type <return> to continue, or q <return> to quit--- select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #4 0x000000000042c218 in run_actions (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1549 t = 0x7f2a75584700 ret = -1 mod = 0x0 ms = 0 __FUNCTION__ = "run_actions" #5 0x000000000041f416 in do_action (h=0x7fffb9fa9e30, a=0x7f2a75585b10, msg=0x7fffb9fa9f10) at action.c:1045 ret = 1 v = 1 dst = {send_sock = 0x7fffb9fa9b50, to = {s = {sa_family = 64710, sa_data = "c\000\000\000\000\000\060\233\372\271\377\177\000"}, sin = {sin_family = 64710, sin_port = 99, sin_addr = {s_addr = 0}, sin_zero = "0\233\372\271\377\177\000"}, sin6 = { sin6_family = 64710, sin6_port = 99, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "0\233\372\271\377\177\000\000Ù¨bu*\177\000", __u6_addr16 = {39728, 47610, 32767, 0, 43225, 30050, 32554, 0}, __u6_addr32 = {3120208688, 32767, 1969400025, 32554}}}, sin6_scope_id = 366}}, id = 10, proto = -24 '\350', send_flags = {f = 234 '\352', blst_imask = 98 'b'}} tmp = 0x1467562aa20 <Address 0x1467562aa20 out of bounds> new_uri = 0x7a0a02 "parse_headers" end = 0x79db5c "core: parser/msg_parser.c" crt = 0x40 <Address 0x40 out of bounds> cmd = 0x7f2a75475010 len = 1967607824 user = 32554 uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x7fffb9fa9b70 "@\234\372\271\004", len = 6529985}, host = { s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, port = {s = 0x0, len = 0}, params = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6661320}, sip_params = {s = 0x7fffb9fa99f0 "", len = 1967607824}, headers = {s = 0x0, len = 1967607824}, port_no = 39488, proto = 47610, type = 32767, flags = 6552774, transport = { s = 0x7fffb9fa9a20 "", len = 1967607824}, ttl = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, user_param = {s = 0x7fffb9fa9a70 "\240\232\372\271\377\177", len = 1956545302}, maddr = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 0}, method = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986448}, lr = {s = 0x1658d8 <Address 0x1658d8 out of bounds>, len = 1790768}, r2 = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}, gr = {s = 0x12e <Address 0x12e out of bounds>, len = 1967607824}, transport_val = {s = 0x14600000000 <Address 0x14600000000 out of bounds>, len = 1969417808}, ttl_val = { s = 0x7fffb9fa9aa0 "\240\233\372\271\377\177", len = 4742114}, user_param_val = {s = 0x0, len = 1967607824}, maddr_val = { s = 0x0, len = 0}, method_val = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6534031}, lr_val = {s = 0x79d630 "core", len = 6552774}, r2_val = {s = 0x7fffb9fa9b00 "n\001", len = 1969417880}, gr_val = {s = 0x7f2a7562aa20 "", len = 1969399988}} next_hop = {user = {s = 0x7fffb9fa9820 "", len = 1967607824}, passwd = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, host = {s = 0x7fffb9fa9870 "\240\230", <incomplete sequence \372\271>, len = 1956545302}, port = { ---Type <return> to continue, or q <return> to quit--- s = 0x1c588e1 "\311\305\001", len = 0}, params = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986216}, sip_params = {s = 0x1659c0 <Address 0x1659c0 out of bounds>, len = 1969416936}, headers = {s = 0x7f2a7562aa20 "", len = 1969399993}, port_no = 301, proto = 0, type = ERROR_URI_T, flags = 0, transport = { s = 0x7f2a7562a8d9 "Content-Type: application/json-rpc\r\nCache-Control: no-cache\r\nPragma: no-cache\r\nUser-Agent: Java/1.8.0_121\r\nHost: \r\nAccept: text/html, image/gif, image/jpeg, *; q=.2, */*; q=.2\r\nConn"..., len = 0}, ttl = { s = 0xb9fa98a0 <Address 0xb9fa98a0 out of bounds>, len = 1969416936}, user_param = {s = 0x0, len = 1967607824}, maddr = { s = 0x0, len = 0}, method = {s = 0x7fffb9fa99a0 "\240\233\372\271\377\177", len = 6534031}, lr = {s = 0x79d630 "core", len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x7fffb9fa9e7c "\260\f\222\017", len = -1174757736}, transport_val = { s = 0x79d630 "core", len = -1174757704}, ttl_val = {s = 0x7a0a10 "get_hdr_field", len = 7986012}, user_param_val = { s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, maddr_val = {s = 0x79d630 "core", len = 1969416864}, method_val = { s = 0x7a0a10 "get_hdr_field", len = 7986012}, lr_val = {s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, r2_val = { s = 0x7f2a75475010 "\001", len = 1969416864}, gr_val = {s = 0x7f2a7562eaa0 <incomplete sequence \350>, len = 1969399988}} u = 0x7f2a7562ee50 port = 0 dst_host = 0x80067e961 i = 0 flags = 0 avp = 0x7f2a75475010 st = {flags = 3120207856, id = 32767, name = {n = 1967607824, s = {s = 0x7f2a75475010 "\001", len = 1969239958}, re = 0x7f2a75475010}, avp = 0x7f2a75475010} sct = 0x4b9fa9c40 sjt = 0x79d630 rve = 0x7f2a75582f20 mct = 0x7fffb9fab700 rv = 0x63fcc6 rv1 = 0x0 c1 = {cache_type = 43, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 7718384}, re = 0x0}, pval = {rs = {s = 0x0, len = 7718384}, ri = 26, flags = 0}}, i2s = "\t\000\000\000\000\000\000\000huTu*\177\000\000\000\000\000\000\000"} s = {s = 0x7fffb9fa99f0 "", len = 1969239512} srevp = {0x3000000018, 0x7fffb9fa9ab0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action"

2 2

unable to receive response from freeswitch
by ysuryas . 13 Mar '17

13 Mar '17

I have configured kamailio (192.168.56.102) and freeswitch(192.168.56.103) in separate virtualboxes when making a call, the invite goes to proxy and it is forwarded to freeswitch but response from freeswitch is "unreachable port" attached the wireshark trace: I don't why 10.0.2.15 is making the Invite to freeswitch, i was expecting 192.168.56.102 to do that. UA -> 192.168.56.102 -> 192.168.56.103 Any suggestions on this? [image: Inline image 1] Thanks, Surya

2 1

rtpproxy doesn't record after start_recording
by Marko Tirs 11 Mar '17

11 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control 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"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

1 0

Dispatcher Module - Hava limit?
by Rodrigo Moreira 11 Mar '17

11 Mar '17

Hello, I am experiencing the following problem: my dispatcher module is losing many calls when the scenario for balancing is about 175 CPS or more. Is there a limit that the module can do? Where do I find this information? Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069

3 6

Multiple parallel branches, one invalid
by Nathan Ward 10 Mar '17

10 Mar '17

Hi, I have a scenario I’m not sure of the best way to solve. I use alias_db and have multiple destinations for a single number, and have append_branches set to 1. I then use lookup_branches to get details from the location table for each branch. I then do t_load_contacts and t_next_contacts, then t_relay to parallel fork the calls to the destination, and all the phones ring at once. This all works great if all the destinations from the dbaliases table are online. However, if one of them is offline, lookup_branches says "Not found in usrloc”. This branch is still there, and the call is attempted. In my environment, this then does a DNS lookup, can’t resolve the domain, and the branch fails. This is the first branch that is tried, and when this branch has an error like this, it causes t_relay() to fail - I check for the output of t_relay() and reply with an error - per the default config that Kamailio ships with. I guess what I need to figure out, is if there’s a way to have t_relay fail only if all branches fail? Or, is there a more elegant way to handle this? I should add, I only allow a single registration per subscriber - so, lookup_branches only returns one branch per alias from alias_db_lookup. -- Nathan Ward

2 1

404 not here for BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/ Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95: F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+ rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. 3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487;branch=z9hG4bK-524287-1--- 1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060;rport=5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

2 1

sip6 BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95:F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;branch=z9hG4bK-524287-1---1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060 ;rport=5060;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

3 2

Bounty for topos fix
by Trevor Peirce 09 Mar '17

09 Mar '17

Hello, I would like to offer a $100 USD bounty if a developer is able to provide a fix for the two topos problems I have experienced. I will add another $50 if this can be solved within 48 hours of the time stamp of this message. The bug report is at: https://github.com/kamailio/kamailio/issues/1005 You may find the most relevant information in my last few updates. Please refer to mar1-nat.txt and mar1-reinvite.txt for the most recent SIP captures, and the post below that for example database records. Steps to reproduce the NAT problem can be found at https://github.com/kamailio/kamailio/issues/1005#issuecomment-283255720 Some fine print to clarify the offer: Payment is contingent on resolving both described use-cases. Payment will be sent via your choice of PayPal, cheque drawn on a US bank, or cheque drawn on a Canadian bank. Payment will be issued once the patch is accepted by the Kamailio project, although the bonus for a quick resolution simply requires a suitable patch to be submitted to the bug tracker within 48 hours from now. Thanks! -- Trevor Peirce AcroVoice Solutions Inc

1 1

Changes in LCR Target List doesn't takes place immediately
by Nitesh Lohchab 09 Mar '17

09 Mar '17

Hello, I believe LCR Target list is stored in the database and any changes made here should affect the behaviour of the calls imedialtely. I have changed one of my Target List Rule , where I changed my gateway Id to another gateway. But the calls are still being routed via previous gateway. Is there something I can do to make the changes in production hours. Sometimes one of carriers delivers bad call quality I would like to switch the carriers. -- Regards, Nitesh Lohchab

2 1

rtpengine sending rtp to wrong endpoint after reinvite
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, One of our customers is using a SEMS box to place two outbound calls using our sip trunk. Once the first call is connected a second call is placed and when the second call answers their server sends a re-invite to switch audio ports so the rtp traffic doesn't flow through their server anymore but is routed inside our platform. Basically, they just switch SDP's of both calls. It seems like a random issue, and is not really reproducible, except for placing multiple calls and sometimes both parties can hear each other, other times they can't, because rtpengine fails (I think) to update the endpoint and keeps sending rtp back to their server for one of the call legs. We tried to reproduce the case using a freeswitch box and it worked every time. After the reinvite, the rtp remained within our platform. The signaling in both cases still goes through the freeswitch or sems for call control. Does anyone have experience with this case? Or seen the issue before where rtpengine keeps sending rtp to the original endpoint? Regards, Grant Bagdasarian CM

2 3

Access avp between sip transactions
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, Is it possible to access an avp variable from a different sip transaction? On an INVITE is set an avp, but on ReINVITE I would like to access this avp again and get the value. Currently the value is null, which makes sense. Not sure if this is possible and if it violates any rules inside the Kamailio engine? I guess I could use a htable, but I'm just wondering if there is a built-in way for doing this. Regards, Grant Bagdasarian CM

2 1

Kamailio Presence help required.
by SamyGo 08 Mar '17

08 Mar '17

Hi, I'm in need of making/tweak an existing Kamailio Presence setup which is giving some tough time. *Whats already working:* BLF dialog states changes are already sent across the users. SCA is working as well. *What isn't working:* When a User comes online then it sends SUBSCRIBE with *Event: dialog* and don't get notified of its subscribers state right then unless the monitored extensions make a call (BLF works) *Why is it not working:* As evident from wireshark traces, the user IP phones (Test sets: Polycoms, Yealink, CISCO, Grandstream) don't send our *Event: presence* rather only *Event:dialog* and Kamailio do not send NOTIFY out to everyone. Though yes there is an internally generated PUBLISH seen and handled properly upon registration state changes. Jitsi has been tested and Kamailio send out these registration state change info to jitsi, somce jitsi sends *Event:presence *in SUBSCRIBE. I need dialog based NOTIFY to be sent out on registration state-change. Need pointers and help on the topic, looking forward to some feedback. Regards, Sammy

4 8

TCP stack is not doing load distribution among children
by Surendra Pullaiah 08 Mar '17

08 Mar '17

Hi Daniel, Thanks for your suggestions. The approaches which you have suggested are good, event I thought before initiating this query (since am working with kamailio for last three years), but my requirement is to distribute on single connection. Since it requires code change, started working on it. Really thanks for understanding my concern. @juha: Dear juha, it is not broken TCP implementation, kamailio TCP stack well implemented. If you would like to build a server to handle HTTP and MSRP protocols at higher rate, here is the place. But for SIP we have options to handle either UDP/TCP. Thanks surendra

1 0

Kamailio push for Android VoIP Application and VoIP Contacts update
by sscc 08 Mar '17

08 Mar '17

good day, i have a running kamailio server having 200 users. I maintain the user ids in a vcf (vCard Format) form. i have developed an Android VoIP application. Android application is given to the user along with the contact vcf. The contact list get increased day by day and consequently the user ids in vcf (vCard Format) as well. the application is also updated. Aim is to update application and the contact list by sending push notification to user. (we don't want to place our application at play store). i want to do push notification from Kamilio server for application and contact list update. Is there any way to do push notification from server to users to update the app and VoIP contact list. regards sscc -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-push-for-Android-VoIP-Appl… Sent from the Users mailing list archive at Nabble.com.

1 0

TCP stack is not doing load distribution among children for SIP
by Surendra Pullaiah 08 Mar '17

08 Mar '17

Dear Daniel, A small concern in kamailio, existing TCP stack is not doing load distribution among children for SIP messages. For example UE-->kamailio1 (TCP listen (4 children)) ----->kamailio2 (TCP Listen 4 children) As per stack if UE opens a connection with kamailio1 based on free children it is sharing the FD, so no problem towards from UE to kamailio1, then need requirement to forward same sip message to kamailio2, here kamailio1 is opening connection with kamailio2 and then forwarding. Suppose if we forward two requests to kamailio2, only one children is getting used among four. With this design suppose if we want to run our kamailio as two sip servers we may not achieve good results when performance matters. Please suggest something on this to move further. Regards surendra

3 5

new keepalive module ?
by Guillaume Bour 07 Mar '17

07 Mar '17

Hi all, I would like to implement destinations keepalive in drouting module (as it is done in dispatcher). But instead of duplicating what's implemented in dispatcher, I think it would be more clever to create a new module dedicated to pinging destinations, and to plug drouting, dispatcher and other modules to this new one What's you opinion about that ? Cheers, Guillaume

1 0

Kamailio core dumped
by Alexey V. Panfilov 07 Mar '17

07 Mar '17

Hi there, I've kamailio 4.2.3 (i386/freebsd) which works perfectly with 200-300 cps for years. Now I need migrate to freebsd amd64. I've installed kamailio 4.4.5 (amd64/freebsd), copied kamailio.cfg, made a test call - it was all right. But when I start work load (200-300 cps), kamailio core dumped and exited. In logs (skipped "<script>:" having): ======================= Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x809937000, 976); Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x809937000, 976) called from core: msg_translator.c: build_req_buf_from_sip_req(2154), module: core; Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [msg_translator.c:2159]: build_req_buf_from_sip_req(): out of memory Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (2/SL) Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x809937000, 864); Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x809937000, 864) called from tm: t_reply.c: relay_reply(1885), module: tm; Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: tm [t_reply.c:1887]: relay_reply(): ERROR: relay_reply: cannot alloc reply shmem Mar 1 12:39:37 sip7 /usr/local/sbin/kamailio[32038]: : <core> [mem/f_malloc.c:598]: fm_free(): BUG: fm_free: bad pointer 0x61743b3e6d6f632e (out of memory block!), called from tm: h_table.c: free_cell_helper(226) - aborting Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: ALERT: <core> [main.c:740]: handle_sigs(): child process 32034 exited by a signal 10 Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: ALERT: <core> [main.c:743]: handle_sigs(): core was generated Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32044]: INFO: <core> [main.c:810]: sig_usr(): signal 15 received ======================= I've tried to start with -m 512 - nothing changes. Hardware: Xeon(R) CPU E5-2660, 16Gb RAM, 1Tb HDD. Software: OS: FreeBSD 10.3-STABLE (r313760) amd64 version: kamailio 4.4.5 (amd64/freebsd) f98162 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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: f98162 compiled on 12:25:48 Mar 1 2017 with clang 3.4 bt is here: http://tmp.lehis.ru/kam_gdb.txt I can't enable debug symbols for gdb (tried gmake mode=debug, which added EXTRA_DEBUG to flags). How can I do it? I need in direction for resolve this issue, please, help. Thanks. -- Best regards, Alexey V. Panfilov

2 3

Kamailio 4.2 crashed
by Igor Potjevlesch 07 Mar '17

07 Mar '17

Hello, I got two coredumps: (gdb) bt full #0 0x000000000061ab8e in qm_insert_free (qm=0x7f0f7fca7000, frag=0x7f0f8f6f6268) at mem/q_malloc.c:180 f = 0x7f0f8f6c45e8 prev = 0x0 hash = 5 And (gdb) bt full #0 0x000000000061b935 in qm_detach_free (qm=0x7f0f7fca7000, frag=0x7f0f8f6d70b8) at mem/q_malloc.c:271 prev = 0x7f0f7fca7248 next = 0x7f0f8f6c45e8 What are necessary to diagnose how this issue occurs? Thank you for your support. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 3

SCTP support for CDP
by Sergey Zyrianov 07 Mar '17

07 Mar '17

Hello, Can I enable use of SCTP for diameter connections by CDP module? Thanks, Sergey.

3 6

Installation of Siremi 4.3.0 on Centos 7.3 ?
by Lucas Viallon 07 Mar '17

07 Mar '17

Hi i want install Siremis 4.3.0 on a Centos 7.3 with Php 7.1 when i want access to the web site, i have this error in httpd logs: Mon Mar 06 18:35:06.833569 2017] [php7:warn] [pid 22018] [client 172.16.1.150:58375] PHP Warning: require_once(SMARTY_CORE_DIRcore.assemble_plugin_filepath.php): failed to open stream: No such file or directory in /var/www/siremis-4.3.0/openbiz/others/Smarty/libs/Smarty.class.php on line 1368 [Mon Mar 06 18:35:06.833641 2017] [php7:error] [pid 22018] [client 172.16.1.150:58375] PHP Fatal error: require_once(): Failed opening required 'SMARTY_CORE_DIRcore.assemble_plugin_filepath.php' (include_path='.:/usr/share/pear:/usr/share/php:/var/www/siremis-4.3.0/openbiz/others/') in /var/www/siremis-4.3.0/openbiz/others/Smarty/libs/Smarty.class.php on line 1368 Anyone know this errors ? regards Lucas

2 1

Error in kamailio (Need for help)
by Ghanshyam Variya 06 Mar '17

06 Mar '17

Mar 2 11:55:33 Ubuntu kamailio: INFO: <core> [core/sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Mar 2 11:55:33 Ubuntu kamailio: INFO: <core> [core/tcp_main.c:4665]: init_tcp(): using epoll_lt as the io watch method (auto detected) Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: rr [rr_mod.c:175]: mod_init(): outbound module not available Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: sipcapture [sipcapture.c:454]: parse_table_names(): INFO: table name:sip_capture Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: <core> [db.c:203]: db_bind_mod(): Module db_mysql not found. Missing loadmodule? Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: sipcapture [sipcapture.c:526]: capture_mode_set_params(): parsing capture_mode: could not bind db funcs for url:[mysql://kamailio:kamailiorw@localhost/kamailio] Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: sipcapture [sipcapture.c:637]: capture_mode_init(): capture mode parsing failed Mar 2 11:55:33 Ubuntu kamailio[6479]: WARNING: sipcapture [sipcapture.c:769]: mod_init(): Default capture mode configuration failed. Suppose sip_capture calls will use other defined capture modes. Mar 2 11:55:33 Ubuntu kamailio[6479]: : <core> [core/mem/q_malloc.c:145]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(0, 0)[0x7fea6ae64ae0:0x7fea6ae64b18]! Memory allocator was called from sipcapture: sipcapture.c:831. Fragment marked by sipcapture: sipcapture.c:788. Mar 2 11:55:33 Ubuntu kamailio: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe This type of error are put's my kamailio so can you tell me why ? Thank you, Ghanshyam variya

3 2

Re: [SR-Users] Unknown $dlg(h_entry) and $dlg(h_id) in dialog
by Никитенко Виталий 06 Mar '17

06 Mar '17

Hi, very thanks! Before use dlg_get i lost 3-4 hour calls every day. Now one day - fly is good! > Hello, > > can you try using dlg_get(...) before the xlog? > > Cheers, > Daniel > > On 27/02/2017 08:33, Никитенко Виталий wrote: >> Hi i have kamailio 4.2, dialog module with parametr >> modparam("dialog", "dlg_flag", 5) >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "profiles_with_value", "caller") >> modparam("dialog", "dlg_match_mode", 1) >> modparam("dialog", "default_timeout", 3600) >> modparam("dialog", "timeout_noreset", 1) >> modparam("dialog", "send_bye", 1) >> modparam("dialog", "timeout_avp", "$avp(i:4242)") >> modparam("dialog", "db_url", DBURL) >> modparam("dialog", "db_mode", 0) >> >> in event_route[tm:local-request] i use >> if(is_known_dlg()) { >> xlog("L_INFO","DIALOG IS KNOWN $dlg(h_entry)::$dlg(h_id)) I >> } >> >> And when call drop by timeout, sometimes in log i see >> /usr/local/sbin/kamailio[21144]: INFO: <script>: DIALOG IS KNOWN <null>::<null> >> >> But usually this string in log file i see as "DIALOG IS KNOWN 3452::3334" >> >> Why it may be? when the $dlg(h_entry)::$dlg(h_id) is not known, I can not track $DLG_lifetime. >> Different between unsuccessful and successful call i dont see >> >> _______________________________________________ >> 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 > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > ------------------------------

2 1

Topoh module & call leg/transaction does not exist
by Olli Attila 06 Mar '17

06 Mar '17

Hello, I recently started testing topology hiding module on Kamailio and encountered a problem where B party hangs up, Kamailio can't identify the ongoing transaction to which the BYE should be related to and instead of tearing the call down Kamailio just sends back "481 Call leg/Transaction does not exist" to B party. On Kamailio config I have only enabled the topoh.so -module with default options and configured this to enable masking: modparam("topoh", "mask_callid", 1) Here is what the INVITE to B party looks like and the BYE message coming from the B party: http://pastebin.com/raw/f63zUmid When debugging on Kamailio end i get: http://pastebin.com/raw/ZefGAz30 Is this a Kamailio issue or a softphone issue on B party end? Cheers, --Olli

3 6

kamailio re.subst question.
by José Seabra 02 Mar '17

02 Mar '17

Hello There, I'm trying using the following regexp on re.subst function: (.*)Calling-Name: ?([\W|0-9A-Za-z_]{5,500} ?[<>:@.|0-9A-Za-z_]{5,500})(.*)$ And the string used for match is the following: #015#012Calling-Name-Status: available#015#012Calling-Name: "josé" < sip:5555555552@10.10.10.10>#015#012Presentation-Indicator: allowed#015#012 The Kamailio re.subst function is constructed as the following: $sht(cnam=>$ci::calling_name)=$(var(rb){re.subst,/(.*)Calling-Name: ?([\W|0-9A-Za-z_]{5,500} ?[<>:@.|0-9A-Za-z_]{5,500})(.*)$/\2/s}); The issue here is that this re.subst function doesn't match the accents, I have tested this regex on regex101.com and it does match. My question is how i can get the same result with re.subst? Anyone can help? Thank you Regards -- José Seabra

2 5

Dimensioning of shared memory for carrierroute
by Igor Potjevlesch 02 Mar '17

02 Mar '17

Hello, I'd like to dimension shared memory to use "kamctl cr reload" without issue. Currently, my table "carrierroute" has 204k entries and if I do "kamctl cr reload", the reload fails. Just after, I have thousands of these errors: Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: <core> [sip_msg_clone.c:507]: sip_msg_shm_clone(): cannot allocate memory Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: tm [t_lookup.c:1332]: new_t(): ERROR: new_t: out of mem: Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: tm [t_lookup.c:1472]: t_newtran(): ERROR: t_newtran: new_t failed After a reboot, Kamailio back to normal. Actually, I start Kamailio with -m 704 and -M 128. Thank you for your advises. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 2

Help with KEEP-ALIVE method
by Sergey Basov 02 Mar '17

02 Mar '17

Hi All. Today I have problem with connection from 1 of the clients. Their PBX sends KEEP-ALIVE after some time after REGISTER. I have next error in kamailio log Feb 28 14:26:19 sbc2 /usr/sbin/kamailio[3657]: ERROR: <core> [tcp_read.c:1354]: tcp_read_req(): bad request, state=7, error=4 buf:#012KEEP-ALIVE sip:1.2.3.4 SIP/2.0#015#012#015#012#012parsed :#012KEEP-ALIVE sip:1.2.3.4 SIP/2.0 After which my kamailio closes TCP session.... This connection is done with sip TLS. Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

3 4

Lost of connection with MySQL?
by Igor Potjevlesch 02 Mar '17

02 Mar '17

Hello, Two times in the last week, Kamailio reports MySQL errors like these: /usr/local/sbin/kamailio[19225]: ERROR: db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Can't connect to MySQL server on '192.168.2.6' (4) (2003) /usr/local/sbin/kamailio[19225]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query /usr/local/sbin/kamailio[19225]: ERROR: acc [acc.c:504]: acc_db_request(): failed to insert into database These errors are repeated many times and cause Kamailio to hang and don't reply to SIP requests. A reboot of Kamailio solves the issue. Anyone has already had this issue and had solved it? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 17

CentOS 6 RPMs don't provide utils.so nor http_client.so
by Trevor Peirce 01 Mar '17

01 Mar '17

Hello, Is there any reason the CentOS 6 RPMs don't build utils.so or http_client.so? Looking back to 4.3 it appears it was being built: kamailio-utils-4.3.6-1.1.x86_64.rpm However it is no longer available in 4.4 nor 5.0 RPMs for either CentOS 6 or 7. Thanks, -- Trevor Peirce AcroVoice Solutions Inc

2 5

Update Kamailio tables
by Andy Chen 01 Mar '17

01 Mar '17

Hi all, What is the quickest way to update Kamailio tables? I wanted to check out the SIPTRACE module and I noticed the table is not there. Thanks. -- Andy Chen

2 2

Kamailio -can we change 478 to 503 when dns fails?
by José Seabra 01 Mar '17

01 Mar '17

Hello there, Some times dns server fails and my kamailio server start replying "Unresolvable destination (478/SL)". The propose of my email is to know if is there any possibility to change this reply (478) to for example one of 5XX replies. I have modparam("sl", "default_code", 503) but in situations where dns fails, kamailio doesn't reply 503. Thank you for your support Regards José Seabra

2 4

Figuring out why topos stops script from working
by Trevor Peirce 01 Mar '17

01 Mar '17

Hello, I'm having some trouble with topos not working well when mixed with nathelper (and maybe other cases, too). There is a bug that describes exactly the problem at https://github.com/kamailio/kamailio/issues/1005 Has anyone else had trouble with this module? Anything I'm missing? Thanks, -- Trevor Peirce AcroVoice Solutions Inc

3 2

Unknown $dlg(h_entry) and $dlg(h_id) in dialog
by Никитенко Виталий 01 Mar '17

01 Mar '17

Hi i have kamailio 4.2, dialog module with parametr modparam("dialog", "dlg_flag", 5) modparam("dialog", "enable_stats", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "timeout_noreset", 1) modparam("dialog", "send_bye", 1) modparam("dialog", "timeout_avp", "$avp(i:4242)") modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 0) in event_route[tm:local-request] i use if(is_known_dlg()) { xlog("L_INFO","DIALOG IS KNOWN $dlg(h_entry)::$dlg(h_id)) I } And when call drop by timeout, sometimes in log i see /usr/local/sbin/kamailio[21144]: INFO: <script>: DIALOG IS KNOWN <null>::<null> But usually this string in log file i see as "DIALOG IS KNOWN 3452::3334" Why it may be? when the $dlg(h_entry)::$dlg(h_id) is not known, I can not track $DLG_lifetime. Different between unsuccessful and successful call i dont see

2 1

Scaling websockets over SIP
by Jade SZ 01 Mar '17

01 Mar '17

Hi Guys, I am using kamailio/rtpengine and webrtc over sip client for a/v calls and all works fine. Now I want to scale it to 2 or more than 2 kamailio servers. As per search on mailing list I could: 1) Websocket Outbound/Edge Proxy 2) Each UA client to register with its Regards, Jade SZ

1 2

pushing alarms/counters other than with SNMPstats
by Valentin Christoph 28 Feb '17

28 Feb '17

Hi all I'm looking for ways to push stats/alarms from kamailio to some external destination other than using the SNMPstats module. Any ideas? Thanks Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

5 5

sip over websockets sip uri
by Jade SZ 28 Feb '17

28 Feb '17

Hi Team, Question: Just wanted to clarify regarding SIP URI in webrtc (over sip) connection. e.g. if we have a scenario where Kamailio is hosted with websocket support. Websocket URI is used to send packets to to wss/ws address and the SIP URI goes with it. I have tested it with any SIP URI including any domain and it works if I authenticate that domain, regardless that domain is real or not. // client side SIP URI config var configuration = { uri : 'sip:alice@example.com', }; Can we use any central domain for sip uri similar to realm concept and send traffic to multiple kamailio servers through websocket LB. dummy.domain - for ws/wss SIP URI kamailio-a.domain.com kamailio-b.domain.com Thanks in advance. Regards, Jade

2 4

Newbie starting - looking for docs / tutorials
by Antony Stone 28 Feb '17

28 Feb '17

Hi. I'm just starting with Kamailio - I've been to a few presentations about it, at Fosdem and KamailioWorld, but never used it so far. I'm pretty familiar with Asterisk, and I want to use them in combination. My first use case is to handle SIP registrations (both as a client, registering to a remote SIP server, and as a server, accepting registrations from handsets) in Kamailio instead of Asterisk, and still be able to route calls to/from the registered accounts through the Asterisk dialplan. I suspect this is a pretty common starting point for people who've used Asterisk and want to add Kamailio to the setup? So far I've been unsuccessful finding any Howto or Tutorial explaining how to do this - I can't help feeling I must simply be looking in the wrong place or in the wrong way. Searching for topics like "Asterisk Kamailio Getting Started" I find: http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf but with 12 chapters and 85 pages I'm not sure which bits are the essential parts to do what I need. I've also found http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x- asterisk-11.3.0-astdb but this seems to focus on a database backend, which I guess I'll want to use in due course, but doesn't seem the simplest place to start just to learn how Asterisk & Kamailio work together? So, please can someone suggest a beginner's guide to introducing Kamailio into an Asterisk setup, keeping things as simple as possible to start with, so I can learn about Kamailio one bit at a time (since there do seem to be a lot of bits to it) from a familiar environment of using it with Asterisk? Thanks in advance, Antony. -- What do you get when you cross a joke with a rhetorical question? Please reply to the list; please *don't* CC me.

2 3

multi domain kamailio
by przeqpiciel 28 Feb '17

28 Feb '17

Let's suppose that i have two machines with installed asterisk and one with kamailio. I would like have routing based on sip domain. for domain sip.domain1.com send sip signalling to asterisk#1 server and for sip.domain2.com send to asterisk#2 server. What i should use? Domain module or what ? Tomasz

3 4

Kamailio v5.0.0 Released - new major version is out
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Kamailio v5.0.0 is out – it comes with 6 new modules and a consistent set of improvements touching more than 50 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. A consistent effort was directed to restructuring the source code tree to meet more modern patterns; introducing the KEMI framework that allows writing SIP routing blocks in other embedded interpreters such as Lua, JavaScript or Python; removal of obsoleted MI control mechanism, being replaced by RPC. Enjoy Kamailio v5.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2017 - http://www.kamailioworld.com

1 0

problem with avpops and mongodb modules
by hdssdsdsdsfsdf hdssdsdsdsfsdf 27 Feb '17

27 Feb '17

1 1

Releasing Kamailio v5.0.0
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Hello, I am considering to release Kamailio v5.0.0 at the beginning of next week, depending how things evolve in the next days, either on Monday, Feb 27, or Tuesday, Feb 28. I haven't noticed any new issues specific for the new features added for 5.0 lately, so it looks pretty good so far. If you are aware of issues not reported to the bug tracker, register them as soon as possible at: - https://github.com/kamailio/kamailio/issues There are still some parts of documentation that have to be updated, I am planning to work on them during the next days, but help from others are very appreciated, especially details of what you had to do changes while upgrading from 4.4 to 5.0. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Parallel Forking with Different Call-Id
by SamyGo 24 Feb '17

24 Feb '17

Hi, I've a very strange scenario to work on which requires me to parallel fork the call to the same Destination provider. The only problem here is that they think that the second INVITE with different branch tag is a re-transmission and hence only take one call forward. I do not have to modify any R-URI or any headers, hence just 'append_branch() before t_relay() is in the code. ... $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); append_branch(); route(RELAY); ... *Question:* Is there anyway possible I can change the CallID of the forked INVITEs ? I have tried using Topoh module, but it still puts the same CallID before sending out. Different from the A-leg but the Sent out INVITEs have same Call-ID value. In other weird scenario, I've also tried branching, and looping call within Kamailio before sending out to carrier hoping that Kamailio would treat the two different calls and Topoh would change the Call-ID on both INVITEs before sending out.. ... if(is_present_hf("X-FORKED")) { $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); route(RELAY); }else { append_hf("X-FORKED: 1\r\n"); append_branch(); route(RELAY); } ... Still I get same Call-ID on outgoing branched call. I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but again I don't want to have that component bottleneck the throughput. That could be my very last option. Looking for some ideas. Regards, Sammy

3 6

Kamailio 4.3.x location_attrs update issue
by Mridul B 24 Feb '17

24 Feb '17

Hi, We've noticed a case where the *location_attrs* table isn't updated when there's an unexpected shutdown of a SIP device (here there'll be no REGISTER message sent with expires=0). We're using Kamailio v4.3.x When the SIP device shuts down ungracefully. The location table is updated using DELETE query This is done by the timer which deletes the expired contacts, by default the timer_interval is set as 60. Here, the logs of auto-removal of expired contacts entries by the timer > Feb 17 17:16:13 localhost kamailio[23273]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:14' AND expires<>'1970-01-01 00:00:00'] > Feb 17 17:16:14 localhost kamailio[23272]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:15' AND expires<>'1970-01-01 00:00:00'] The delete operation is only on location table, shouldn't we have the same implementation on the location_attrs table based upon ruid or username? We're getting duplicate entries in the location_attrs table of the same sip user after it registers. But when the SIP device un-registers, the entries in the location as well as in the location_attrs table is deleted Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: usrloc [ucontact.c:1725]: > uldb_delete_attrs(): trying to delete location attributes > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location_attrs where username='mridul' > AND ruid='uloc-58a72243-5ace-2'] > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location where username='mridul' AND > contact='sip:mridul@127.0.0.1:37881;rinstance=b841a8c916e47dfa'] I think the usrloc module should have the implementation of deleting the entries of expired sip users based on its ruid/username in the location_attrs table also. Thanks for the help. Best regards, MRIDUL

2 3

want to see active users SIP users
by Mandar Nagarkar 24 Feb '17

24 Feb '17

Hello Everyone, I want to see active users SIP users for this purpose I checked the location table of kamailio database it does have active users entries however as per my observation there is 45 secs to 1 min delay before the entry actually added to this table i.e. suppose if I am registering the user through pjspi,jssip or any client I cannot see entry for that user in location table. I have to wait up-to 1 min to see the user in location table. Is there any way we can Identify if user is online without delayed entry? Regards, Mandar nagarkar The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 1

Fetching proxy-autheticate header from 407 Response
by vivek＠advaitamtech.com 23 Feb '17

23 Feb '17

Hi All, Am not using the authentication in the kamailio. So I will be receiving the "407 Proxy Authentication Required" response from the third party server for the first INVITE. I want to fetch the Proxy-Authenticate header present in the 407 response. I browsed and tried using the pseudo-variables "$adu/$aa/$ar". It dint work. Please let me know if there is a way to fetch this header. Thanks, Vivek.

3 4

How to release call without having to use fromTag and toTag, and resume a suspended call
by Cassandra Yurekli 23 Feb '17

23 Feb '17

To whom it may concern, There are several points I don’t quite understand regarding a couple of kamailio modules. I have a main application which controls calls by sending command messages via a socket (for example reject this sip invites). I communicate with the socket client via evapi module and using evapi_asynch_relay() in order to be sure that communication are suspended and kamailio waiting an order for doing something with them (Accepte, reject, release, ...). I want to release a call, using a command “release this call”. I’m actually doing this with dialog module (using get_dialog(callid,fromTag,toTag) and dlg_bye(‘all’)) but is there anything else (like another module) which could release a call without necessarily having to use fromTag and toTag to get the context ? I didn’t find it on modules documentations. Also after an INVITE I’m waiting for the main app to accept or reject the call. For accepting and continuing the transaction I found “t_continue(tindex,tlabel,route)” on tmx module, but at this moment $T(id_index) and $T(id_label) are null, is it normal or I did something wrong ? How can I simply resume the suspended call ? Thanks; Best regards,

2 1

Need help with tcpops
by Abdoul Osséni 22 Feb '17

22 Feb '17

Hi, My call flow is: Softphone --> TLS ---> Kamailio --> UDP --> ASTERISK --> PSTN I want to use tcpops module between the softphone and kamailio. http://www.kamailio.org/docs/modules/4.4.x/modules/tcpops - enable tcpops : no problem - disable tcpops : two cases - cancel or bye from softphone: no problem. - cancel or bye to softphone via kamailio: how disable tcpops? I can't use $avp(bye_conid) because it is asterisk thant sending the sip message. Can we use $avp(caller_conid)? In my case, the softphone is always at the origin of the call establishment. Regards Abdoul.

2 1

Automated integration tests for individual modules
by A Lai 22 Feb '17

22 Feb '17

Hi, I just started working with the Kamailio code base. I am wondering if there are some common strategies in having automated tests on proper functioning of modules when loaded into cfg.

2 1

problem with avpops and mongodb modules
by hsdsdssdfsdf dsasd 22 Feb '17

22 Feb '17

2 1

How to handle CANCEL if corresponding INVITE transaction is missing
by Venkatesh Macha 22 Feb '17

22 Feb '17

Hi all, I am using Kamailio 4.4.5, It is working fine. But in one case. If i make a video call between the two users and cancels the call then Call is not disconnecting on the callee side. Please look the following scenario. 1000 made one video call to 2000 2000 is ringing. Now 1000 disconnected the call by sending CANCEL. But 2000 is not disconnecting. This is happening because Kamailio is unable to find the corresponding INVITE transaction for the CANCEL so it is simply dropping the requests. Why Kamailio unable to find the transaction. And is there any possibility to handle this case and send the CANCEL to callee side. I tried with forwad() and someother things but it is just generating the loops. I am also attaching the ngrep trace. Thank you in Advance. CANCEL_not_working_nabble.CANCEL_not_working_nabble <http://sip-router.1086192.n5.nabble.com/file/n156243/CANCEL_not_working_nab…> -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-handle-CANCEL-if-correspondi… Sent from the Users mailing list archive at Nabble.com.

2 2

kamailio memory leak in private memory
by david 22 Feb '17

22 Feb '17

Hello all We are observing an issue in one kamailio proxy we use. The consecuence of an unknow cause is that the private memory of some UDP receiver processes in where we listen from a particular IP, seems to be draining along the time while processing traffic. Quite "slowly" but maybe aroung 200K in 3-4hours, depending on the moment of the day. This is quite weird for us, because it seemed like it was caused by some particular traffic profile, since we have several IP listening (8 children each) and we only see the issue in the UDP receiver processes of some of those IPs. Besides we have moved the kamailio service listening on the same IP to several different physical servers, and the behaviour is the same. We look at it ussing pkg.stats kamcmd command and for instance if we track one of the process affected pid: 23784 rank: 53 used: 2838616 free: 13005408 real_used: 3771808 pid: 23784 rank: 53 used: 2842584 free: 12999280 real_used: 3777936 This 2 outputs were taken with 10min diff between each other. But we dont know where or how to look at it. We have been these last days trying to search for patterns to allocate some particular traffic source causing something weird, but no luck. We have checked all the SIP INVITE messages looking for anything weird at any field but no luck, besides we sometimes see some bad headers with blank spaces in the uri or thing like that but those cases we have an error and we dont think it can cause a memory leak. Shared memory seems to be stable. We also have checked the config we are using and we have been discarding possible issues since we dont think the $var variables we are using could cause that issue (but you never know....) We tried to debug the kamailio on one affected IP compiling it with DBG_SYS_MALLOC , and doing the dump of the pkg mem status when the kamailio was restarted and no traffic was processed, and also when it had traffic processed. in this last case, we removed traffic, waited for 20min and made the dump i attach you both files, we have the debug logs, but they are very big to be passed here, and we dont know what to look for :) could you please give us some guidance or clue to at least start to searching for something that can cause this issue? thanks alot and regards david

3 6

parse_addr_spec(): unexpected char [^M] in status 6:
by Abdoul Osséni 21 Feb '17

21 Feb '17

Hello, Could you help us to fix this error? <core> [parser/parse_addr_spec.c:662]: parse_addr_spec(): unexpected char [^M] in status 6: Regards Abdoul.

3 5

Transaction-persistent data and uac_req_send()
by Alex Balashov 21 Feb '17

21 Feb '17

Hello, I am using uac_req_send() to send INVITEs to a redirect server. However, upon receiving the redirect, I need access to some state information that was last available at the time of sending the request. Does uac_req_send() allow me to persist AVPs like a normal TM transaction? If not, what are my options, other than building my own state using htable or what have you? I would really rather not maintain—and be responsible for garbage-collecting—externally constructed state like that if possible. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 10

Kamailio 4.4 with 2 Asterisk 13
by Graham Brown 20 Feb '17

20 Feb '17

I'm trying to load balance 2 Asterisk servers with Kamailio offloading the registration to Kamailio using the Asterisk real-time database tables. I have a problem with the dispatcher module, if there is a t_on_failure route it always goes into that, if I remove that then I would expect every call to round robin but I'm only ever hitting the same server from the same extension (seems like sticky session based on source ip). 1. how do I fix my failure route so I can reboot server 1 and server 2 will be used 2. how to I change the sticky session behaviour. Another issue I am facing with extensions registering to Asterisk are only avaliable on the server they routed to My assumption was if using the Asterisk real-time database I would get the contact on both servers. Maybe my architecture is incorrect, I'm open to suggestions. Currently Kamailio is not storing locations (assuming that is the extension contact) maybe that is a better solution, but I dont know how to configure Asterisk to forward everything to Kamailio then or integrate via the database or something. Thanks in advance for any help Regards Graham

2 1

Cleaning wiki accounts
by Daniel-Constantin Mierla 20 Feb '17

20 Feb '17

Hello, I noticed a lot of random usernames for the accounts created to edit the wiki. Apart of several which did some which did some edits to add spam, most of them are inactive. Not sure what would be the real purpose behind them, but I am going to remove anything that doesn't look as a community user. Shall I remove an account with random id, but for a good user, let me know, it can be readded/account recreated. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

memory: Free fragment not found
by Dmitri Savolainen 17 Feb '17

17 Feb '17

Hi. My kamailio v4.4.5 installation ( -m 512 -M 64 ) stop sip processing with repeatable error: Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:415]: > fm_search_defrag(): fm_search_defrag(0x7f39917fa010, 952); Free fragment > not found! > Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:498]: > fm_malloc(): fm_malloc(0x7f39917fa010, 952) called from core: rvalue.c: > rval_new_empty(224), module: > core; Free fragment not foun while it was enough free shared memory according to kamcmd> core.shmmem m > { > total: 512 > free: 472 > used: 27 > real_used: 39 > max_used: 51m > fragments: 37035 > } whether it was about package memory or smth else? Thanks! -- Savolainen Dmitri

2 4

MSILO offline messages are coming in reverse order.
by Venkatesh Macha 17 Feb '17

17 Feb '17

Hi all, I am using MSILO module for the offline messaging feature. It is working good. But If I send more than one message to the offline user. Those messages are delivering in the reverse order. Please consider the following scenario. A sending B ( B is offline, so messages are stored in DB) hi hello how are you Now B comes online and registers. Kamailio sending the messages but App is getting them in reverse order. how are you hello hi Is there any possibility to tell Kamailio to send them in order. ? Thank you in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/MSILO-offline-messages-are-coming-i… Sent from the Users mailing list archive at Nabble.com.

2 2

Problem with dispatcher on Kamailio 4.2 and Debian 8
by przeqpiciel 17 Feb '17

17 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 4

Kamailio 4.2 on Debian 8 and dispatcher problem
by przeqpiciel 16 Feb '17

16 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 1

Re: [SR-Users] Timejump causes kamailio to hang/stall
by erik 16 Feb '17

16 Feb '17

Thanks Daniel, I will look into that. Indeed I found the logic that corrects the time ticks for any drifting, using the time of day. I guess that causes the registrations to expire. I am out of the office for a couple of days, but will get back to you. Best Regards, Erik Hello, On 15/02/2017 14:47, Erik Visser wrote: > After I have kamailio up and running, with UAs properly registrered and sessions routed successfully, my platform faces a timejump. > The platform starts up with date&time set to 1-1-2000 and after kamailio is up and running, the date&time jumps to the current date&time, causing two effects: > - I loose all my registrations (since they expired in 3600 seconds) > - kamailio does not respond to any SIP messages for a couple of minutes > > Is there a way to have kamailio use timeticks rather than relative time for the registrations? > Does anybody recognize the hanging behavior? Any suggestions? > kamailio is using unixtime for registrations, not relative time. But the timer callbacks are using relative intervals to know when to be executed (e.g., timer callbacks are defined to be executed like very 1 minute, or every 90 seconds). It may be that some of the timers try to catch up on missed intervals. It is strange that you say no sip traffic is processed. Can you simulate again the situation and get the backtrace with gdb from few sip workers? If your traffic is over udp, then do 'kamctl ps', get the pid of processes with index 1, 2 and 3, then use gdb to attach to those PIDs and grab the output of: bt full info locals list Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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remove offline message prefix of Offline message of MSILO module
by Venkatesh Macha 16 Feb '17

16 Feb '17

Hi all, I am implementing Offline messages functionality using the MSILO module. Everything went well. I am able to store and send those offline messages when the user comes online. But my present problem is, I don't want the offline message prefix. Please look at the following example. This is what Kamailio sending. [Offline message - Wed Feb 15 05:43:38 2017] Test2 I want only Test2 Is there any possibility to remove that offline message prefix ? Thank you in advance ! -- View this message in context: http://sip-router.1086192.n5.nabble.com/remove-offline-message-prefix-of-Of… Sent from the Users mailing list archive at Nabble.com.

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