- sr-users - kamailio.org

Set a system variable in the request route
by Nicolas Breuer 20 Oct '18

20 Oct '18

Hello, How can I change a system variable within the route ? I can change a variable witk kamctl but any ideas in the code ? Thanks

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Variable Kamailio version
by Nicolas Breuer 20 Oct '18

20 Oct '18

Hello, Can we get the Kamailio version into a variable ? Thanks

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pause/waiting Timer kamailio
by eyas barhouk 19 Oct '18

19 Oct '18

Hello Dears if i need to make kamailio able to pause request processing for a Period of time is that possible ? for example when kamailio receive an INVITE request then it have to wait until 10 seconds before relay it to the second server. thanks in advance .

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Acoounting with acc_radius when dialog terminates by default_timeout
by Denys Pozniak 19 Oct '18

19 Oct '18

Hello! How can I send RADIUS Accounting-Request in case of terminating dialog via default_timeout timer? I tried to catch Kamailio's own generated BYE in event_route AND tried to use cdr_expired_dlg_enable , but it looks like acc module is not triggered. event_route[dialog:end] { if ( dlg_isflagset(FLT_ACC_START) && is_method("BYE") ) { setflag(FLT_ACC); } dlg_resetflag(FLT_ACC_START); } modparam("acc", "log_flag", FLT_ACC) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACCMISSED) modparam("dialog", "default_timeout", 3600) modparam("dialog", "send_bye", 1) version: kamailio 4.4.2 (x86_64/linux) 892ad6 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 12:58:38 Aug 28 2017 with gcc 4.8.5 -- BR, Denys Pozniak

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Help
by Olivier KOFFI 19 Oct '18

19 Oct '18

Good morning I want to use kamailio to send calls to an asterisk. My problem is that I want to define call prefixes. example If the subscriber calls the 6000, please call us 7000 on transfer to asterisk. Subscribers are at 6000.6001, etc. I want to know exactly which line to modify in kamailio.cfg thank you Sincerely KOFFI Koffi Olivier +225 79830005 +225 59098608 Directeur Technique LIFOR Sarl

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Event route websocket:closed and function unregister
by Володимир Іванець 19 Oct '18

19 Oct '18

Hello all! I have *event_route[websocket:closed]* configured. By using *$si*, *$sp* variables and data from special hash table I can figure out information about account that was disconnected. I'm trying to use *unregister* function against but with no luck. 1. I'm creating variable which contains (as an example): " sip:100-test@127.0.0.1" string, where "100-test" is username. 2. lookup("location", "$var(user_uri)") - runs fine. I'm getting a match and lots of useful variables including *$ruid* are being set: */usr/sbin/kamailio[27169]: DEBUG: registrar [lookup.c:256]: lookup_helper(): contact for [100-test] found by address* 3. But unfortenately *unregister* with same variable does not work. unregister("location", "$var(user_uri)"): *DEBUG: <core> [core/parser/parse_uri.c:1282]: parse_uri(): bad uri, state 0 parsed: <you@> (4) / <you(a)kamailio.org <you(a)kamailio.org>> (16)* *ERROR: <core> [core/parser/parse_to.c:81]: parse_to_uri(): failed to parse To uri* *ERROR: registrar [save.c:1040]: unregister(): failed to extract Address Of Record* I was trying to manipulate *$var(user_uri)* variable's content but could not make it work. It's interesting that variant with *$ruid* - *unregister("location", "", "$ruid")* will work but unfortenately it works only with *modparam("usrloc", "db_mode", 1)*: *ERROR: usrloc [urecord.c:655]: delete_urecord_by_ruid(): delete_urecord_by_ruid currently available only in db_mode=3* I was wonderring if someone could tell me if I'm just using *unregister* function with uri parameter in wrong format or there might be some other problem. Thanks a lot!

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Dispatcher Return
by Duarte Rocha 19 Oct '18

19 Oct '18

Greetings, I'm using Kamailio with Dispatcher and now i have the need to have more than one destination with the same IP:Port belonging to the same Destination Group. Those destinations will have different configurations unrelated to the Dispatcher. However, after ds_select_dst I only have access to IP and Port. Is it possible to Dispatcher to return another fields from the same table row? Or return any other way to identify the row. Thank you for your help. Best Regards, Duarte Rocha

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Questions
by Olivier KOFFI 19 Oct '18

19 Oct '18

Bonjour Je veux utiliser kamailio pour transmettre des appels vers asterisk. Mon problème est que je veux définir les prefixes d'appels. exemple Si l’abonné appel le 6000 kamailio gère l'appel.mais s'il compose le 7000 on transfert à asterisk. Les abonné sont au 6000,6001,etc je veux savoir exactement quelle ligne modifier dans kamailio.cfg merci Cordialement KOFFI Koffi Olivier +225 79830005 +225 59098608 Directeur Technique LIFOR Sarl

2 1

HTTP Post with JSON
by eyas barhouk 18 Oct '18

18 Oct '18

Hello Dears i'm trying to send POST HTTP request with JSON content on Kamailio V 5.2.0. i have two main headers "content type" & "Authentication " in addition i have the json content . when i'm using curl request it look like : Curl -H "Authentication ket = xxxxx" -H "content type : JSON" -d " {the JSON content}" https://xxxxx.com/send So how can i use the Http_client module to send such requests ?? thanks in advance .

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Http_async_client stops script execution
by Denys Pozniak 18 Oct '18

18 Oct '18

Hello! Please help me to understand how http_async_client module behaves. I am going to use it for call routing according to the http response. if (is_method("INVITE")) { t_newtran(); xlog("L_ERR", "ASYNC0-$fU-$rU-$rm---- \n"); http_set_timeout(1000); http_set_method("POST"); http_append_header("Content-Type: application/json"); http_async_query("http://10.10.10.10:8080/RPC", "{\"jsonrpc\": \"2.0\", \"method\": \"ul.lookup\", \"params\": [\"location\",\"user\"], \"id\": 1}", "HTTP_REPLY"); xlog("L_ERR", "ASYNC1-$fU-$rU-$rm---- \n"); } route[HTTP_REPLY] { xlog("L_ERR", "HTTP_REPLY: ok $http_ok \n"); xlog("L_ERR", "HTTP_REPLY: status $http_rs\n"); xlog("L_ERR", "HTTP_REPLY: body $http_rb\n"); xlog("L_ERR", "HTTP_REPLY: error $http_err)\n"); } In syslog I see HTTP_REPLY log strings with correct response, but do not see ASYNC1 string and script execution stops. What's wrong with this? # kamailio -v version: kamailio 4.4.2 (x86_64/linux) 892ad6 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 12:58:38 Aug 28 2017 with gcc 4.8.5 -- BR, Denys Pozniak

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No way to send SIP reply from app_perl or jsonrpc modules
by M S 18 Oct '18

18 Oct '18

Hi, I have a use case where I need to send a statefull (or stateless) reply to non-INVITE SIP transaction from PERL script using app_perl module's "moduleFunction" method, https://kamailio.org/docs/modules/5.0.x/modules/app_perl.html#ID-7b0efd72c8… It fails with error "function not found, missing load module?", even though the documentation says that at least "sl_send_reply" method should work. As an alternative, I tried to use JSONRPCS module to call RPC methods "tm.reply" and "tm.reply_callid" from PERL script (using libcurl), https://kamailio.org/docs/modules/5.0.x/modules/jsonrpcs.html https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.rpc.reply https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.rpc.reply_callid However, they only seems to work for INVITE transactions. If I try to use them for e.g. MESSAGE request I get "Request Timeout" on libcurl and "500 I am terribly sorry" on SIP client. Are these bugs or suppose to work this way? If so, then, is there any way i can send a SIP reply to non-INVITE SIP transaction from outside of the kamailio.cfg script? Please help. P.S. I am using Kamailio 5.0.x on debian wheezy 32bit virtual machine. Thank you.

1 0

Script @Startup
by Nicolas Breuer 17 Oct '18

17 Oct '18

Hello All, Is there a way to execute some codes/scripts at the launch of the Kamailio Server ? So only once when we launch the server. Thanks

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Re: [SR-Users] Missing documentation ims_usrloc_scscf module
by Pan Christensen 17 Oct '18

17 Oct '18

Thanks Henning and Carsten. It should also be noted that the module is required by both IMS_ISC and IMS_REGISTRAR_SCSCF. The lacking documentation of IMS_USRLOC_SCSCF also make those modules difficult to implement. Pan B. Christensen Developer Phonect AS > -----Original Message----- > From: Henning Westerholt <hw(a)kamailio.org> > Sent: tirsdag 16. oktober 2018 20:27 > To: sr-users(a)lists.kamailio.org > Cc: Carsten Bock <carsten(a)ng-voice.com>; Pan Christensen > <pan.christensen(a)phonect.no> > Subject: *** SPAM *** Re: [SR-Users] Missing documentation > ims_usrloc_scscf module > > Am Dienstag, 16. Oktober 2018, 13:31:08 CEST schrieb Carsten Bock: > > can you open a ticket on the GitHub Tracker for this? Otherwise this > > might get lost at some point... I don't have time to take care of it > > immediately as I am travelling this and next week, more likely I would > > take care of it later in November. > > Hello Carsten, > > i have opened already a ticket about that one month ago. It is already > assigned to ngvoice. ;-) > > https://github.com/kamailio/kamailio/issues/1644 > > Best regards, > > Henning > > -- > Henning Westerholt - https://skalatan.de/blog/ Kamailio security assessment > - https://skalatan.de/de/assessment

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Sharing xavp’s between nodes.
by Igor Olhovskiy 17 Oct '18

17 Oct '18

Hi! Is there any possibility to share xavp’s between Kamailio nodes? Idea is to share result of ds_select_dst between several nodes and as 5.2 Kamailio stores result in xavp - need to share this structure as well Right now sharing between nodes is done on sht + dmq and I’m not sure I can save xavp in sht. Regards, Igor

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kamcli v1.1.0 released
by Daniel-Constantin Mierla 17 Oct '18

17 Oct '18

Hello, kamcli tool just got its first official release, a step forward in the process of packaging it. It is a command line tool that aims to be a modern alternative to the venerable kamctl, being written in Python and having already a large set of implemented sub-commands, such as: * subscriber – manage SIP subscribers * ul – manage user location records * address – manage permissions address records * aliasdb – manage database aliases * db – manage kamailio database content * dialog – manage active calls (dialog) * dialplan – manage dialplan records * dispatcher – manage load balancer (dispatcher) * group – manage group membership records (acl) * moni – continuous refresh of the values for a list of statistics * mtree – manage memory trees (mtree) * ps – print the details for kamailio running processes * rpc – interact with kamailio via jsonrpc control commands (alias of jsonrpc) * rpcmethods – return the list of available RPC methods (commands) * speeddial – manage speed dial records * srv – server management commands (sockets, aliases, …) * stats – get kamailio internal statistics * tls – management commands for TLS profiles and connections * uptime – print the uptime for kamailio instance The news article for this release is available at: * https://www.kamailio.org/w/2018/10/kamcli-v1-1-0-released/ The corresponding tag in the github repository being: * https://github.com/kamailio/kamcli/releases/tag/v1.1.0 How to install kamcli and examples of usage can be found at: * https://github.com/kamailio/kamcli/blob/v1.1/README.md Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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Kamailio topoh and t_check_trans not working.
by José Seabra 17 Oct '18

17 Oct '18

Hi all, I noticed that using the module TOPOH with the parameter mask call id enabled and the t_check_trans together isn't working correctly. When i receive the BYE message my script call the function t_check_trans() in order to check if a message belongs or is related to a transaction. This function always returns false when used with the topoh mask call id enabled, once that the BYE belongs to an existing dialog it must return true. In the logs it says something like transaction doesn't match dialog using the RFC3261. If the TOPOH mask call id option is disabled everything works perfectly. Anyone here already faced this behavior? Thanks Regards -- José Seabra

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Missing documentation
by Pan Christensen 16 Oct '18

16 Oct '18

Hello. The documentation for the ims_usrloc_scscf module seems to be missing: Not Found The requested URL /docs/modules/stable/modules/ims_usrloc_scscf.html was not found on this server. ________________________________ Apache/2.4.10 (Debian) Server at www.kamailio.org Port 443 Pan B. Christensen Utvikler Phonect AS Mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> + 47 41 88 88 00 [mail_footer]

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Re: [SR-Users] Problem with callid as htable key
by Daniel-Constantin Mierla 16 Oct '18

16 Oct '18

If no operator is matched in the first two characters of expression part for $shtcn(...), then regexp is used. That's because initially the $shtcn() was doing only regexp matching, later it was added support for additional operators. Yes, == is the operator for equal. Cheers, Daniel On 16.10.18 11:53, jenus(a)cyberchaos.nl wrote: > Daniel, > > Ok so if i'm correct i can prefix the callid with for example 1234 to > avoid matching the eq. In other words eq will only match the regex > code if the callid string starts with it? > > And by mathing on equal you mean: $shtcn(ht_test=>==$var(htid)) ? > > Thanks, > > Jan > > Daniel-Constantin Mierla schreef op 2018-10-16 10:16: >> Just to add that regexp matching when using keys from the sip message >> such as call-id is exposed further to some issues -- for example, it is >> common to have . (dot) in call-id, which for a regexp is matching any >> char, so a call id having 123.456 is matching 123x456. So depending on >> what you want to count, might be better to match by equal or prefix >> rules. >> >> Cheers, >> Daniel >> >> On 16.10.18 10:03, jenus(a)cyberchaos.nl wrote: >>> Daniel, >>> >>> Thanks for the quick response, that indeed solves my problem. >>> >>> >>> Jan >>> >>> Daniel-Constantin Mierla schreef op 2018-10-16 09:55: >>>> Hello, >>>> >>>> the expression in the $shtcn(table=>exp) can have some operators at >>>> the >>>> beginning, see: >>>> >>>> - >>>> https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#shtcn_htable_… >>>> >>>> >>>> >>>> So, eq is one of them, the safest is to explicitly set the operator if >>>> you cannot predict what is the key, like in this case. If the operator >>>> is missing, then the exp is matched as regular expression, so you can >>>> do: >>>> >>>> $shtcn(ht_test=>~~$var(htid)) >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 16.10.18 09:47, jenus(a)cyberchaos.nl wrote: >>>>> Hello, >>>>> >>>>> I'm running kamailio 5.1.4 and running into a issue with htable. I >>>>> use >>>>> it to store some dialog data so i can use it in a http route. When i >>>>> use the callid as a key it creates a issue for callid's starting with >>>>> eq. Here some example code that i use to reproduce the issue: >>>>> >>>>> >>>>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>>>> >>>>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>>>> + $Ts; >>>>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>>>> eq1234"); >>>>> $var(htid) = "eq1234"; >>>>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>>>> $var(htable-val) = $null; >>>>> >>>>> if ($shtcn(ht_test=>$var(htid)) > 0) >>>>> { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>>>> } else { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>>>> } >>>>> $var(htid) = $null; >>>>> >>>>> Results in : >>>>> >>>>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>>>> HT_TEST: Adding 531::926::1539675391 to ht_test with key eq1234 >>>>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>>>> HT_TEST: Key eq1234 NOT found! >>>>> >>>>> >>>>> When i prefix eq with a 1 it works fine: >>>>> >>>>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>>>> >>>>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>>>> + $Ts; >>>>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>>>> eq1234"); >>>>> $var(htid) = "1eq1234"; >>>>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>>>> $var(htable-val) = $null; >>>>> >>>>> if ($shtcn(ht_test=>$var(htid)) > 0) >>>>> { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>>>> } else { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>>>> } >>>>> $var(htid) = $null; >>>>> >>>>> Results in: >>>>> >>>>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>>>> HT_TEST: Adding 3050::1098::1539675445 to ht_test with key eq1234 >>>>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>>>> HT_TEST: Key eq1234 found! >>>>> >>>>> >>>>> Am i running into a bug? Or are there some restrictions on the >>>>> key's i >>>>> can use? I'm still going trough the logs to check if there are more >>>>> callid's that have a issue with matching. >>>>> >>>>> >>>>> Thanks, >>>>> >>>>> Jan Hazenberg >>>>> >>>>> _______________________________________________ >>>>> Kamailio (SER) - Users Mailing List >>>>> sr-users(a)lists.kamailio.org >>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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Re: [SR-Users] Problem with callid as htable key
by Daniel-Constantin Mierla 16 Oct '18

16 Oct '18

Just to add that regexp matching when using keys from the sip message such as call-id is exposed further to some issues -- for example, it is common to have . (dot) in call-id, which for a regexp is matching any char, so a call id having 123.456 is matching 123x456. So depending on what you want to count, might be better to match by equal or prefix rules. Cheers, Daniel On 16.10.18 10:03, jenus(a)cyberchaos.nl wrote: > Daniel, > > Thanks for the quick response, that indeed solves my problem. > > > Jan > > Daniel-Constantin Mierla schreef op 2018-10-16 09:55: >> Hello, >> >> the expression in the $shtcn(table=>exp) can have some operators at the >> beginning, see: >> >> - >> https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#shtcn_htable_… >> >> >> So, eq is one of them, the safest is to explicitly set the operator if >> you cannot predict what is the key, like in this case. If the operator >> is missing, then the exp is matched as regular expression, so you can >> do: >> >> $shtcn(ht_test=>~~$var(htid)) >> >> Cheers, >> Daniel >> >> >> On 16.10.18 09:47, jenus(a)cyberchaos.nl wrote: >>> Hello, >>> >>> I'm running kamailio 5.1.4 and running into a issue with htable. I use >>> it to store some dialog data so i can use it in a http route. When i >>> use the callid as a key it creates a issue for callid's starting with >>> eq. Here some example code that i use to reproduce the issue: >>> >>> >>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>> >>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>> + $Ts; >>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>> eq1234"); >>> $var(htid) = "eq1234"; >>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>> $var(htable-val) = $null; >>> >>> if ($shtcn(ht_test=>$var(htid)) > 0) >>> { >>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>> } else { >>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>> } >>> $var(htid) = $null; >>> >>> Results in : >>> >>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>> HT_TEST: Adding 531::926::1539675391 to ht_test with key eq1234 >>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>> HT_TEST: Key eq1234 NOT found! >>> >>> >>> When i prefix eq with a 1 it works fine: >>> >>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>> >>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>> + $Ts; >>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>> eq1234"); >>> $var(htid) = "1eq1234"; >>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>> $var(htable-val) = $null; >>> >>> if ($shtcn(ht_test=>$var(htid)) > 0) >>> { >>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>> } else { >>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>> } >>> $var(htid) = $null; >>> >>> Results in: >>> >>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>> HT_TEST: Adding 3050::1098::1539675445 to ht_test with key eq1234 >>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>> HT_TEST: Key eq1234 found! >>> >>> >>> Am i running into a bug? Or are there some restrictions on the key's i >>> can use? I'm still going trough the logs to check if there are more >>> callid's that have a issue with matching. >>> >>> >>> Thanks, >>> >>> Jan Hazenberg >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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Problem with callid as htable key
by jenus＠cyberchaos.nl 16 Oct '18

16 Oct '18

Hello, I'm running kamailio 5.1.4 and running into a issue with htable. I use it to store some dialog data so i can use it in a http route. When i use the callid as a key it creates a issue for callid's starting with eq. Here some example code that i use to reproduce the issue: modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" + $Ts; xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key eq1234"); $var(htid) = "eq1234"; $sht(ht_test=>$var(htid)) = $var(htable-val); $var(htable-val) = $null; if ($shtcn(ht_test=>$var(htid)) > 0) { xlog("L_WARN", "HT_TEST: Key eq1234 found!"); } else { xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); } $var(htid) = $null; Results in : Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: HT_TEST: Adding 531::926::1539675391 to ht_test with key eq1234 Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: HT_TEST: Key eq1234 NOT found! When i prefix eq with a 1 it works fine: modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" + $Ts; xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key eq1234"); $var(htid) = "1eq1234"; $sht(ht_test=>$var(htid)) = $var(htable-val); $var(htable-val) = $null; if ($shtcn(ht_test=>$var(htid)) > 0) { xlog("L_WARN", "HT_TEST: Key eq1234 found!"); } else { xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); } $var(htid) = $null; Results in: Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: HT_TEST: Adding 3050::1098::1539675445 to ht_test with key eq1234 Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: HT_TEST: Key eq1234 found! Am i running into a bug? Or are there some restrictions on the key's i can use? I'm still going trough the logs to check if there are more callid's that have a issue with matching. Thanks, Jan Hazenberg

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Problem with Kamailio 4.2.0
by José Antonio Gutiérrez Delgado 16 Oct '18

16 Oct '18

Hi, I'm using Kamailio 4.2.0 and I'm developing a SIP-based application. Today I made a test connecting two devices directly to an S7 sharing the Internet, I usually use a VPN even if they are both connected to the same network because in the future they will be in different networks.Without the VPN calls were not received. This has never happened to me, I have repeated tests and in an S6 it has also happened to me. So I have checked my Kamailio server and I have detected that when executing: kamctl ul show, the received field did not appear: With VPN: Contact:: sip:pruebaT@47.60.42.40:10924;transport=udp Q= Expires:: 1992 Callid:: iwNoYb~QR- Cseq:: 21 User-agent:: Unknown (belle-sip/1.6.1) Received:: sip:47.60.42.40:10924 State:: CS_NEW Flags:: 0 Cflag:: 64 Socket:: udp:10.0.0.4:5060 Methods:: 4294967295 Ruid:: uloc-5a9d9097-9fa-431c Instance:: <urn:uuid:8b50dde3-d07f-4752-9a24-7b8ddf219a99> Reg-Id:: 0 Last-Keepalive:: 1539614732 Last-Modified:: 1539614732 Without VPN: Contact:: sip:pruebaT@37.10.156.189:51163;transport=udp Q= Expires:: 1996 Callid:: webjvHycBs Cseq:: 21 User-agent:: Unknown (belle-sip/1.6.1) State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:10.0.0.4:5060 Methods:: 4294967295 Ruid:: uloc-5a9d9097-9fc-1feb Instance:: <urn:uuid:89306a81-e2d1-4b7a-8206-107e3cbc0688> Reg-Id:: 0 Last-Keepalive:: 1539614531 Last-Modified:: 1539614531 Would someone know why it happened? Thanks -- José Antonio Gutiérrez Delgado Responsable técnico Oficina: (+34) 923040031 <http://www.arsoft-company.com/> *Este mensaje se dirige exclusivamente a su destinatario/s y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción.* *Antes de imprimir, piensa si es necesario. Cuidar el medioambiente es tarea de todos.*

3 2

Hi guys,
by shailendra kumar 15 Oct '18

15 Oct '18

I am fresher and want to learn about Kamailio Soft switch, Anyone can provide me link or document from where i can learn. I will be thankful for such help, Thanks & Regards, Shailendra Kumar

2 1

Subscription from asterisk
by willy 15 Oct '18

15 Oct '18

Hello all. I'm trying to share states and presence from kamailio between asterisks. Is there a way to do this in kamailio? Thank you, willy d

2 1

SDP in fifth 200 OK is not the same as in first four
by Roman Romanchenko 15 Oct '18

15 Oct '18

Hi all, I have an issue with SDP from Kamailio side in the following scenario: simple call from A to B. When B answers a call and sends '200 OK' to A, A delays ACK for a while, so '200 OK' re-transmission happens. After the fourth try to re-send '200 OK' towards A, Kamailio changes SDP (specifically connection information 'c', IP address become address of A instead of proxy IP as it was in the previous attempts). At some point A sends ACK and call is established, but last received '200 OK' has invalid SDP and it prevents successful media establishment. I wonder if there any re-transmission counter or whatever what can affect this scenario and change SDP? Roman Romanchenko VoIP devel Sipwise GmbH

2 1

Dialog variables and persistence
by Alex Balashov 15 Oct '18

15 Oct '18

Hi, I forget, can $dlg_var(...) variables nowadays be reconstituted, along with dialog state in general, using database backing? What about profile membership? Last time I brushed with this topic was probably in 4.x days, and as I remember the general idea was that dialog state could be recovered on restart, but without dialog variables or profile counts. Is this accurate? Is it still true? Or has the amount of state serialised into database persistence increased since then? Thanks much! -- 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/

2 1

UPDATE relayed from wrong socket
by Sergiu Pojoga 15 Oct '18

15 Oct '18

Hi ppl, I have this problem with call transfer, may be someone can help. The phone to the far right is registered with the Registrar to the far left using two PATH headers (trespassing two proxy ports, 5070 then 5060). As you can see in the graph below, after receiving the UPDATE request, Kamailio relays it further from port 5060, I expect it to be from 5070 just like the dialog forming INVITE and the CANCEL afterwards. [image: image.png] The UPDATE has a to-tag, but unlike the original INVITE - it has no Route header!??? route[*WITHINDLG*] { if (!has_totag()) return; if (loose_route()) { route(DLGURI); if (is_method("BYE")) { ... } else if ( is_method("ACK") ) { route(NATMANAGE); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { ... } # handle UPDATE method for in-dialog requests if (is_method("*UPDATE*")) { route(DLGURI); record_route(); route(RELAY); } } Thanks in advance.

2 7

Distorted uri / Route header replaces the uri
by Abdulaziz Alghosh 15 Oct '18

15 Oct '18

Greetings, in order to upgrade kamailio server from 3.0.3 to 5.1.4 version, I am conducting test calls and it is not completelly successful. The main components I have are: 1- Media Gateway (Asterisk) : 192.168.226.133 2- Signalling Gateway (Kamailio 5.1.4) 192.168.226.132 3- Global Dispatcher (Kamailio 3.0.3 in front of the Signalling Gateway.) : 192.168.226.130 Both caller and callee have the dispatcher's IP as a proxy and their registration requests is passed to be registered at the Signalling Gateway (Kamailio 5.1.4) and saved on the MySQL server. The caller reaches the callee properly, who in its turn answeres the call and sends the OK msg to the caller who gets it. Unfortunately, when the caller sends the ACK for this INVITE to the callee, the ACK does not reach the callee because it is trapped by the dispatcher at the last stage because the uri was dropped and the Route header was used instead. This behaviour is abnormal since as I know the Route header should be converted into a Via header and the uri stays untouchable. The ACK was looped into the dispatcher itself .i.e. the dispatcher sent it to itself adding always the via header ""Via: SIP/2.0/UDP 192.168.226.130;branch=0"" endlessly. Once again, here is an example of the ACK before the dispatcher and how the dispatcher forwards the ACK afterwards. the supposed path is as following: caller --> GDP--> SGW --> MGW--> SGW --> GDP --> callee Although the GDP (before the calle after the SGW) received the uri well from SGW inside the ACK as it appears below: ACK sip:yyyyyyyyyyyy@192.168.226.130:5060;rinstance=1608f7c153f40093;transport=UDP SIP/2.0 Record-Route: <sip:192.168.226.132;lr=on;ftag=as3b989c5f> Via: SIP/2.0/UDP 192.168.226.132;branch=z9hG4bK136f.1e8349c235fff01ee1849f3456f8868a.0 Via: SIP/2.0/UDP 192.168.226.133:5060;rport=5060;branch=z9hG4bK34871fe9 Route: <sip:192.168.226.130;lr;ftag=as3b989c5f;did=d5b.87890dc6> Max-Forwards: 69 From: "xxxxxxxxxxxxxx" <sip:xxxxxxxxxxxx@domain.com>;tag=as3b989c5f To: <sip:yyyyyyyyyyyyy@192.168.226.132:5060>;tag=1af2c424 Contact: <sip:xxxxxxxxx@192.168.226.133:5060> Call-ID: 7ed2ab533d4c43086746d0e620bdff06(a)domain.com CSeq: 102 ACK User-Agent: MGW Content-Length: 0 The global dipstacher (GDP) has used the Route header (which has no R-URI yyyyyyyyy) instead of the uri: ACK sip:192.168.226.130;lr;ftag=as3b989c5f;did=d5b.87890dc6 SIP/2.0 Record-Route: <sip:192.168.226.132;lr=on;ftag=as3b989c5f> Via: SIP/2.0/UDP 192.168.226.130;branch=0 Via: SIP/2.0/UDP 192.168.226.132;rport=5060;branch=z9hG4bK136f.1e8349c235fff01ee1849f3456f8868a.0 Via: SIP/2.0/UDP 192.168.226.133:5060;rport=5060;branch=z9hG4bK34871fe9 Max-Forwards: 68 From: "xxxxxxxxxx" <sip:xxxxxxxxxxx@domain.com>;tag=as3b989c5f To: <sip:yyyyyyyyy@192.168.226.132:5060>;tag=1af2c424 Contact: <sip:xxxxxxx@192.168.226.132:5060> Call-ID: 7ed2ab533d4c43086746d0e620bdff06(a)domain.com CSeq: 102 ACK User-Agent: MGW Content-Length: 0 Thanks in advance for your contribution Abdulaziz

2 1

DIALPLAN Module: doubts about regular expression usage
by Stefano Bertuola 15 Oct '18

15 Oct '18

Hi Experts. I did some tests with DIALPLAN module and I have some doubts about how to define the regular expressions in the database. For example, using the start character '^' (beginning of the string), it looks not working: kamcmd> dialplan.dump 1 { DPID: 1 ENTRIES: { ENTRY: { PRIO: 1 MATCHOP: 2 MATCHEXP: ^(\+39)(.*)$ MATCHLEN: 0 SUBSTEXP: ^(\+39)(.*)$ REPLEXP: \2 ATTRS: 1 } } } kamcmd> dialplan.translate 1 "+39123456" error: 500 - No translation Removing it, it works... but not as desired: kamcmd> dialplan.dump 1 { DPID: 1 ENTRIES: { ENTRY: { PRIO: 1 MATCHOP: 1 MATCHEXP: (\+39)(.*)$ MATCHLEN: 0 SUBSTEXP: (\+39)(.*)$ REPLEXP: \2 ATTRS: 1 } } } kamcmd> dialplan.translate 1 "+39123456" { Output: 123456" Attributes: 1 } kamcmd> dialplan.translate 1 "xxxxxxx+39123456" { Output: 123456" Attributes: 1 } Can someone help me understand how the regular expression should be used in DIALPLAN, please? Br. Stefano

4 8

DIALPLAN Module: how to get/use the Attributes value
by Stefano Bertuola 14 Oct '18

14 Oct '18

Hi experts. I intended ATTRIBUTES is a value that is returned by dialplan's dp_translate as the result as shown by kamcmd command: kamcmd> dialplan.translate 2 s:+3912345 { Output: 3912345 Attributes: test } Is my understanding correct? I tried many ways, including using AVP and XAVP for the result, but no success: dp_translate("2", "$rU/$xavp(dp_result)"); pv_xavp_print(); DEBUG: dialplan [dialplan.c:222]: dp_get_ivalue(): dpid is 2 from constant argument DEBUG: dialplan [dialplan.c:243]: dp_get_svalue(): searching 7 DEBUG: dialplan [dialplan.c:331]: dp_translate_f(): input is +3912345 DEBUG: dialplan [dp_repl.c:592]: translate(): regex operator testing over [+3912345] DEBUG: dialplan [dp_repl.c:662]: translate(): found a matching rule 0x7f56aef690d8: pr 1, match_exp ^\+39..... DEBUG: dialplan [dp_repl.c:481]: rule_translate(): copying match <3912345> token size 2 DEBUG: dialplan [dialplan.c:340]: dp_translate_f(): input +3912345 with dpid 2 => output 3912345 INFO: <core> [xavp.c:494]: xavp_print_list_content(): +++++ start XAVP list: 0x7f56aeffaaa8 (level=0) INFO: <core> [xavp.c:498]: xavp_print_list_content(): *** XAVP name: dp_result INFO: <core> [xavp.c:499]: xavp_print_list_content(): XAVP id: 3539756229 INFO: <core> [xavp.c:500]: xavp_print_list_content(): XAVP value type: 2 INFO: <core> [xavp.c:509]: xavp_print_list_content(): XAVP value: 3912345 INFO: <core> [xavp.c:531]: xavp_print_list_content(): ----- end XAVP list: 0x7f56aeffaaa8 (level=0) If possible, how can I get both OUTPUT and ATTRIBUTES as result? Stefano

2 1

db_mysql when compiling Kamailio
by Wilkins, Steve 12 Oct '18

12 Oct '18

Hello All, When db_mysql is selected in the make, the make looks for -lmariadb. However, I want MySQL, not mariadb. Is there a way to let the make know that MySQL is preferred ov mariadb? Thank you

2 1

registrar timer and database modes
by eyas barhouk 11 Oct '18

11 Oct '18

hello dears , i'm using kamailio 5.0.1 as IMS platform and have the following issues : * for Module "ims_usrloc_scscf" there is no documentation and i can't find a way to set database mode =2 and define the tables . * for Module "ims_registrar_scscf"when the user Unregistered it takes about 1~2 minutes to remove it from the memory on database mode 0 . so is there a way to minimize the timer when the user Unregistered and remove it immediately from the memory . thanks in advance & best regards.

2 1

Kamailio with Asterisk as Media Relay
by Benjamin Marty 11 Oct '18

11 Oct '18

Hello Ok, that sounds perfect. Is there a sample kamailio configuration file of how to achieve such a thing?

3 2

Addition of a code of conduct to the Kamailio repository
by Henning Westerholt 11 Oct '18

11 Oct '18

Hello, with commit a213d9d1 I introduced a Code of Conduct to the Kamailio repository [1]. This topic was initially discussed during the Kamailio development workshop, started from Daniel. During the development workshop we came to the conclusion that this is a policy decision and needs broader discussion in the Kamailio Management board and also the Kamailio Admin-Team. This discussion started last Friday, and after several days of discussion we decided to choose an existing Code of Conduct. We adapted the CoC of the Debian Project [2] and did a few changes for readability and also to adapt it to the (smaller) scope of our project. This CoC is in my opinion a good, generic and pragmatic document. Many people in our community are already in one way or the other involved in the Debian project. I am aware that this is (especially right now) a hot topic for some people. My personal opinion is that we managed to get a great fit to the existing Kamailio community culture. The history of this project shows that we are an open project and welcome contributions from anybody. So I don't expect any impact because of this change. We also ensured that the CoC only applies inside the Kamailio project, and that the Kamailio Management board is the authoritative instance in any conflicts about this topic. I don't want to start a classical "bike-shedding" discussion on this list about this topic.. But if you have any questions about this, lets us know. You can also contact the Kamailio Management Board, if you want to keep it private. Best regards, Henning Westerholt [1] https://github.com/kamailio/kamailio/blob/master/CODE_OF_CONDUCT.md [2] https://www.debian.org/code_of_conduct

1 0

Websocket configuration for kamailio 5.1.6
by Priyaranjan Nayak 09 Oct '18

09 Oct '18

Hi All, I have compiled the kamailio v5.1.6 source code with tls and websocket module enabled in ubuntu 18.04 LTS. Now I am able to start the kamailio in UDP and TCP port 5060. I wanted to run the kamailio in WS, WSS and TLS port as well. I have generated the private_key, certificate and ca_list and provided the absolute file path in tls.cfg file. I observed tls.so and websocket.so is present in the module path after compile and installation of code. I have used below command to compile and install. 1. make proper 2. make PREFIX="/usr/local/kamailio-5.1" include_modules="db_mysql tls websocket" cfg 3. make all 4. make install I have attached the kamailio.cfg and tls.cfg. Could you please suggest the modification in configuration file to run in WS, WSS and TLS port ? Thanks Priyaranjan

2 1

Kamailio 5.1.6 Debian Package -> SEGFAULT
by Floimair Florian 09 Oct '18

09 Oct '18

Hi! I just upgrade from 5.1.5 to 5.1.6 on my machine and I see Kamailio segfaulting (SIGSEGV). Unfortunately the systemd service is also not restarted in this case. I will generate coredumps and look into the backtrace and provide you with any insights I can gather.

1 0

force_send_socket
by David Villasmil 08 Oct '18

08 Oct '18

Hello all, I'm very rusty when it comes to variables :( There's a part of my config where i load a parameter from usr_preferences to know whether i should append a branch and call him to a fixed IP as well as to their current location. I tried doing this on the REQUEST_ROUTE and append a branch like: # This will check if the TO user is an office user if (avp_db_load("$to", "$avp(office)")) { if ( $avp(gsm)==1 ) { xlog("L_ERR","[REQUEST_ROUTE]: [$rm] callid: $ci from=$fU to=$tU should be ($avp(gsm)) branched - Appending branch\n"); append_branch("sip:$tU@1.2.3.4:5060"); } } And later when the script hits the RELAY, the branch is actually sent, BUT from the *PRIVATE IP.* I tried adding: force_send_socket(67.215.186.219:5060); Right after the "append_branch" to set the socket from where to send, but kamailio is still sending from the private ip. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 11

Kamailio with Asterisk as Media Relay
by Benjamin Marty 08 Oct '18

08 Oct '18

Hello I would like to ask if there is any available informations on how to use Kamailio as a SIP Proxy but letting (multiple) Asterisk Instances do the Media Relaying. Is this scenario supported and if yes are there any tutorials or documentation available for it? I know the Kamailio Dispatcher Module, but this does as far as I know a dispatching of also the SIP Traffic. Not just the Media Traffic.

4 4

set_advertised_address with variable
by Patrick Wakano 08 Oct '18

08 Oct '18

Hello list, Hope you all doing fine! I am using version 5.0.7 and I was trying to use the set_advertised_address with an avp variable but it does not replace with the avp value.... I then find out this thread https://lists.kamailio.org/pipermail/sr-users/2016-March/092327.html, in which is mentioned: *"I expect those functions not to support pvs, but I think this issomething to address in 5.0 and get all core functions working with pvs."* Looks like this was actually not done.... is that correct? Is there any plan to do it? In my case I want to use the set_advertised_address after reading the value from DB, so I can't use the advertised_address or listen options.... Any idea on how I could achieve that? Thank you, Kind regards, Patrick Wakano

3 6

Sending INVITE from loopback subnet
by Joonas Kamailio 08 Oct '18

08 Oct '18

Dear List, I am trying to achieve a scenario where I can recognize the original peer and mask it as peer from subnet 10.0.7.0/24 with kamailio. Basic IP connectivity is established with the following network configuration (Debian9): auto lo:1 iface lo:1 inet static address 10.0.0.0/24 And of course I have routed 10.0.0.0/24 to servers public IP address (hereby known as x.x.x.1) in my router. Please consider the following scenario: - Peer1 192.168.0.1 is sending INVITE to kamailio at IP x.x.x.1 - Kamailio sends INVITE to peer2 at 172.10.0.1 from socket 10.0.0.1 - Peer2 receives the INVITE and knows that since it came from IP 10.0.0.1 it means that INVITE is originally from peer1 - IF peer2 sends INVITE to kamailio, it should be sent to peerX from socket 10.0.0.2 This works if I list both 10.0.0.1 and 10.0.0.2 with 'listen=' parameter but things get out of hand if I have multiple peers. In my tests I have noticed that 60 peers equals 60 listen parameters which equals roughly 500 kamailio processes. At some point there are just too many processes and everything melts. If I omit the listen parameter altogether, kamailio has only one socket from /24 subnet and it is 10.0.0.0. Error message when listen parameter is omitted: [pv_core.c:2612]: pv_set_force_sock(): no socket found to match [10.0.0.1] My kamailio has the following configuration in request_route: $avp(e_private) = $(sht(get_private=>$si)); $fs = $avp(e_private); I know that the case is a bit confusing and totally something normal people won't do. If you have any questions I am happy to answer. Best Regards, Joonas Keskitalo

2 1

Server_id expects number only
by Denys Pozniak 08 Oct '18

08 Oct '18

Hello! I need to separate Kamailio's dmq_usrloc by server_id and will be much easily if server_id can accept string values also, like: server_id="192.168.100.100" But currently it accepts only integer: https://www.kamailio.org/wiki/cookbooks/5.1.x/core#server_id Syslog: 0(11488) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/global.cfg, line 126, column 11-24: syntax error 0(11488) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/global.cfg, line 126, column 11-24: number expected -- BR, Denys Pozniak

2 4

High number of aborted mysql connections
by Sergiu Pojoga 08 Oct '18

08 Oct '18

Hi all, I'm trying to understand why some Kamailio processes become Connor MacLeod and go to live forever causing high # of aborted connections. While watching in time over *SHOW PROCESSLIST*, some 12 kamailio connections in *SLEEP* state, most of them are reused over time (time value gets reset). However, a select few go on to SLEEP forever for as much as the mysql's *wait_timeout* allows them to. Default was 28800 sec, that's how long they lived. If I lower this to say 600 sec, even higher abort rate is observed. I suppose my questions are: 1. Is this normal to have so many SLEEPing connections? 2. What is the recommended MySQL *wait_timeout* value under Kamailio? 3. Any other suggestions as to why such a high rate of aborted connections? Merci!

2 2

Intercepting local reply attributes
by Alex Balashov 06 Oct '18

06 Oct '18

Hi, I have Kamailio in a weird interop/adaptation layer role with the following call flow: SUBSCRIBE to Kamailio with Expires: 0 ---> <-- immediate 200 OK from Kamailio [some time passes] <-- NOTIFY with additional info from Kamailio 200 OK to Kamailio ---> The NOTIFY follows a series of lengthy async HTTP interrogations, so it's constructed at a later point in time with $uac_req(...). The problem is that the standard which regulates this is extremely eccentric in the following way: despite the immediate expiration of the subscription, the NOTIFY is expected to conform to the format of an in-dialog request as if a dialog existed. Don't ask. It's as idiotic as it sounds. But it's a Broadsoft "standard". I can match the To URI / To-tag by capturing it from the incoming From header on the SUBSCRIBE and reproducing it into the outgoing To header on the NOTIFY. The issue is that my From-tag on the NOTIFY does not match the To-tag on the 200 OK for the SUBSCRIBE. So, from the point of view of the subscriber, the NOTIFY is "From" a different party to the one that "initiated" a "dialog" in response to its subscription. I'm just sending the "200 OK" for the SUBSCRIBE like this: --- append_to_reply("Event: $hdr(Event)\r\n"); append_to_reply("Expires: $hdr(Expires)\r\n"); append_to_reply("Contact: <sip:$sel(cfg_get.domain.sip_domain):$Rp>\r\n"); send_reply("200", "OK"); --- I am not aware of anything like an onsend_route that I can use to gain access to the To header of this reply. Nor am I aware of any means of removing headers from a reply and injecting my own To header with a self-generated To-tag I can later mimic in the From tag of the NOTIFY. Any creative workarounds would be appreciated! -- 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/

1 0

kamailio + rtpproxy behind NAT
by Bonjour Madame 05 Oct '18

05 Oct '18

Hello, I have kamailio and rtpproxy running on the same server located behind a router with NAT just like below. UA(internet)---public ip---router(NAT)--Kamailio+RTPproxy(private ip)--UA(private network) There are several issues like one way audio. I need users on the internet to communicate with each other, users on the private network to communicate with each other, and users on the internet and the private network to communicate with each other. What is the best approach? Best regards, Clarence

3 5

SIP communication does not reach phone when STUN is used
by Kevin Olbrich 05 Oct '18

05 Oct '18

Hi! I noticed a minor problem when I set up the STUN config of my phone. When the server is set, I am unable to reach the endpoint from kamailio (Phone1 -> Kamailio -> Asterisk -> same Kamilio -> no outgoing SIP flow to Phone2). If I do not set it, it seems to fallback to it's RFC1918 address which is then detected by nat_uac_test() and I can reach the phone. During my tests with Asterisk directly, this did not make any difference. Seems like Asterisk checks more than FLAG 19/8 in nat_uac_test(). If I am right, I strictly used the SIP source port for answers instead of any self-detection using STUN or similar. My kamailio node is connected to the internet directly and 95 percent of (planned) phones are behind random NATs. In example kamailio.cfg there are these settings: # Caller NAT detection > route[NATDETECT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (is_method("REGISTER")) { > fix_nated_register(); > } else { > if(is_first_hop()) { > set_contact_alias(); > } > } > setflag(FLT_NATS); > } > #!endif > return; > } AND (modified for direct RTP flow): # RTPProxy control and signaling updates for NAT traversal > route[NATMANAGE] { > #!ifdef WITH_NAT > if (is_request()) { > if(has_totag()) { > if(check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > } > } > if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; > if(nat_uac_test("8")) { > fix_nated_sdp("15"); > if (is_request()) { > if (!has_totag()) { > if(t_is_branch_route()) { > add_rr_param(";nat=yes"); > } > } > } > if (is_reply()) { > if(isbflagset(FLB_NATB)) { > if(is_first_hop()) > set_contact_alias(); > } > } > #!endif > return; > } I checked the docs but I am not sure which FLAGs are safe to use (or break things). Any thoughts or hints are welcome. Thank you very much! Kind regards Kevin

2 1

Re: [SR-Users] Kamailio as TCP UDP proxy
by M K 05 Oct '18

05 Oct '18

Thanks, that was very helpful. But now I get an "ERROR: <core> [tcp_main.c:4070]: handle_tcpconn_ev(): connect <Client IP>:<Port> failed" displayed when trying to register. The SIP client establishes the connection to the proxy via TCP, but does not receive an answer. Could this indicate a faulty NAT-ALG?

2 1

clang compiler warnings
by Juha Heinanen 05 Oct '18

05 Oct '18

I just made a test and, for sure, clang warns about unused variables: $ clang -Wunused-variable -I/usr/include/re -I/usr/include/baresip -lbaresip -lrem -lre baresip.c baresip.c:48:6: warning: unused variable 'unused' [-Wunused-variable] int unused; ^ 1 warning generated. -- Juha

3 3

Kamailio memory manager
by Igor Olhovskiy 05 Oct '18

05 Oct '18

Hi! What is the best way to debug messages like... 29(42) ERROR: [core/mem/q_malloc.c:290]: qm_find_free(): qm_find_free(0x7fad301cf010, 56); Free fragment not found! 29(42) ERROR: [core/mem/q_malloc.c:423]: qm_malloc(): qm_malloc(0x7fad301cf010, 56) called from core: core/data_lump.c: dup_lump_list_r(537), module: core; Free fragment not found! 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Cause core.shmmem shows total: 536870912 free: 522859104 used: 12224144 real_used: 14011808 max_used: 14479048 fragments: 10560 Also mod.stats all pkg Module: core { init_io_wait(475): 6960 init_io_wait(524): 6312 new_db_id(272): 512 dupl_string_name(72): 32 db_do_init2(298): 488 mk_switch_cond_table(43): 144 sr_wtimer_add(330): 48 as_asciiz(161): 24 init_modules(1026): 16 rpc_hash_add(146): 2048 db_allocate_columns(150): 32 db_new_result(114): 56 get_hdr_field(116): 1160 db_allocate_columns(160): 16 cnt_hash_add(383): 1024 register_select_table(458): 48 add_callback(59): 320 sr_wtimer_init(308): 136 cfg_new_group(79): 704 cfg_declare(50): 3360 fix_sock_str(420): 480 init_dst_set(83): 32208 grp_hash_add(234): 808 fix_hostname(1301): 152 add_alias(91): 464 add_alias(93): 256 dupl_string(48): 176 parse_headers(325): 432 subst_parser(291): 144 subst_parser(274): 64 exp_optimize_right(411): 24 fix_expr(541): 600 mk_case_stm(3775): 400 fixup_regexp_null(213): 72 mk_elem(90): 560 route_new_list(200): 1344 fix_param(1196): 1336 parse_select(204): 1488 fix_param(1257): 536 fix_match_rve(3028): 320 mk_rval_expr1(2616): 142656 tr_new(1646): 1464 fixup_pvar_all(279): 1624 fix_param(1162): 16144 mk_action(118): 350168 mk_rval_expr2(2674): 489552 mk_rval_expr_v(2537): 1475600 yyparse(2757): 8792 pv_parse_format(1150): 89064 pv_cache_add(347): 31456 parse_params2(585): 1400 set_mod_param_regex(110): 376 set_mod_param_regex(121): 832 tr_table_add(1875): 400 sr_cmd_exports_convert(235): 36288 register_module(280): 6624 load_module(496): 2640 new_sock_info(235): 224 fix_socket_list(1509): 48 new_sock_info(230): 3920 add_alias(93): 56 add_alias(91): 96 yyparse(1774): 64 yyparse(667): 152 yyparse(2306): 128 yyparse(648): 144 sr_push_yy_state(1666): 32 sr_push_yy_state(1656): 104 sr_push_yy_state(1596): 184 addstr(1413): 415752 rpc_hash_add(100): 25656 str_hash_alloc(59): 512 pv_table_add(236): 17864 pv_init_buffer(2055): 327680 pv_init_buffer(2045): 320 init_nonsip_hooks(43): 24 init_rlist(146): 8 mk_rval_expr_v(2548): 1264 route_add(124): 5872 str_hash_alloc(59): 768 rval_get_str(1256): 56632 pp_define(1788): 2248 init_counters(122): 128 cnt_hash_add(332): 20608 str_hash_alloc(59): 1280 Total: 3602152 } Module: rtpengine { child_init(1812): 256 Total: 256 } Module: db_mysql { db_mysql_new_connection(65): 96 db_mysql_new_connection(75): 3040 Total: 3136 } Module: siptrace { mod_init(344): 384 Total: 384 } Module: sl { sl_register_callback(460): 88 Total: 88 } Module: acc { parse_acc_extra(116): 1512 Total: 1512 } Module: dialog { dlg_bridge_init_hdrs(66): 80 Total: 80 } Module: rr { register_rrcb(61): 80 Total: 80 } Module: db_text { dbt_get_columns(67): 64 Total: 64 } Module: permissions { get_pathname(242): 64 Total: 64 } Module: auth { generate_random_secret(239): 32 generate_random_secret(238): 32 Total: 64 } Module: xlog { mod_init(214): 4104 xdbg_fixup_helper(500): 6032 xlog_fixup_helper(535): 26992 Total: 37128 } Module: dmq { make_socket_str_from_uri(149): 24 Total: 24 } Module: sqlops { sqlops_tr_buffer_init(46): 2048 sql_init_con(83): 240 Total: 2288 } Module: textops { tr_txt_parse_re(212): 32 hname_fixup(3217): 768 fixup_method(3287): 1384 Total: 2184 } Module: pv { tr_parse_string(2343): 120 tr_parse_string(2332): 32 tr_parse_string(2378): 96 tr_parse_string(2368): 184 tr_parse_string(2544): 576 add_var(58): 3616 add_var(65): 1112 Total: 5736 } Module: textopsx { fixup_hname_param(598): 352 Total: 352 } Module: avpops { avpops_parse_pvar(57): 416 Total: 416 } Module: userblacklist { check_blacklist_fixup(522): 16 Total: 16 } Module: htable { fixup_ht_key(348): 176 pv_parse_ht_name(158): 520 Total: 696 } Module: dialplan { dp_trans_fixup(396): 296 Total: 296 } Module: dispatcher { Total: 0 } Module: tmx { Total: 0 } Module: tm { Total: 0 } Module: kex { Total: 0 } Module: usrloc { Total: 0 } Module: topos { Total: 0 } Module: utils { Total: 0 } Module: app_python { Total: 0 } Module: cfgutils { Total: 0 } Module: nathelper { Total: 0 } Where is the place to look at? Means what module can cause this? Regards, Igor

2 3

Astricon: OpenSource Soiree
by Giovanni Maruzzelli 05 Oct '18

05 Oct '18

Hello friends and collegues, A nice get together has been organized for evening Wednesday 10th at 7pm The idea is to have drinks and chitchat about opensource telephony and webrtc, and all opensource project founders, members, and practitioners are invited. Please join me, as distinguished participants too! Check it out at http://party.officering.com (no animals has been harmed in this mail) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 2

Kamailio as TCP UDP proxy
by M K 05 Oct '18

05 Oct '18

Hi, I'm struggling with the configuration to use Kamailio as TCP UDP proxy: [SIP Client] <-(TCP)-> [Proxy] <-(UDP)-> [SIP Server] should be the setup. I only have one client and one SIP server. How do I get Kamailio configured as a proxy so that translation between TCP and UDP works (and vice versa)? I have not yet found any documentation and examples that have caused messages to arrive at my SIP server. I think I've been on the wrong track so far and I'm looking forward to any help. Thank you in advance, Laqoore

3 2

Development for Kamailio v5.2 series is frozen
by Daniel-Constantin Mierla 05 Oct '18

05 Oct '18

Hello, short note to mark the freezing of development for Kamailio v5.2 series. For few weeks, please do not push new features in the master branch. Once the branch 5.2 is created (expected to happen in 3-4 weeks from now), the master branch becomes again open for new feature. Meanwhile the focus has to be on testing current code. Work on related tools (e.g., kamctl) or documentation can still be done as well as getting the new modules in 5.2 in good shape, plus adding exports to kemi interface (which should not interfere with old code). The entire testing phase is expected to be 4 to 6 weeks, then the release of v5.2.0 will be out. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

PATH when used with multiple listening ports
by Sergiu Pojoga 04 Oct '18

04 Oct '18

Hi again, When acting as an intermediary proxy for a non-colocated Registrar, the REGISTER request coming in on one port and going out on another, shouldn't the *add_path[received]()* generate *two* nodes in the Path header? Otherwise, how are the subsequent requests supposed to be routed back to the UAC? Scenario with Kamailio proxy on IP 1.1.1.1 with dual port listeners UAC => proxy:5070 <=> proxy:5060 => Registrar:5060 Path: <sip:1.1.1.1;lr;received=sip:123.123.123.123:59593> I would expect it to be something like: Path: <sip:1.1.1.1;lr>,<sip:1.1.1.1:5070;lr;received=sip: 123.123.123.123:59593> modparam("rr", "enable_double_rr", 1) modparam("path", "use_received", 1) Merci!

2 2

Kamailio in a mobile world - ts_append issue
by Dan Quinney 04 Oct '18

04 Oct '18

Using "Kamailio in a mobile world" as a base, I have the following routes to store information about the current transaction, however, if the transaction is already cancelled registrations for the user will still trigger `ts_append`, raising an error as the htable hasn't expired yet. Is there anyway to remove the htable entry when the original transaction cancels? Or something else that I'm missing?

2 1

Kamailio v5.1.6 Released
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, Kamailio SIP Server v5.1.6 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.6. For more details about version 5.1.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/10/kamailio-v5-1-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Roadmap to next major release - v5.2.0
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, as it was discussed during our last IRC devel meeting before the summer, the time to release a new major version should be this autumn. With the Kamailio Developers Meeting planned for September 27-28, 2018, I think it is better to freeze the code couple of days after that event. So I propose the following milestones: * on Thursday, October 4, 2018 - last development day, code for 5.2.0 will be frozen at the end of the day * 4 to 6 weeks of testing * release sometime in November, to be decided more precisely after the first weeks of testing, some rough candidates being within the weeks of Nov 5 or Nov 19 If anyone has something to add or adjust, let's discuss on sr-users mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

2 3

Planing the release of Kamailio v5.1.6
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, a matter of the schedule to be agreed in the morning and the availability during the Kamailio Developers Meeting, we may do a new stable release from branch 5.1.6 on Thursday, Sep 27, 2018, in the afternoon. Given that several of us will be in the same room, the special goal would be to try to automate the process as much as possible. If the time doesn't allow it, then the release will be done next week, a specific day to be decided by the end of this week. Anyhow, this is the informal notification to report as soon as possible any issues you are aware of and not yet in bug tracker, so they get a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

2 4

tcp Free fragment not found
by Diego Nadares 04 Oct '18

04 Oct '18

Hi guys! We have four a scripts that sends dlg.list command to kamailio through xmlrpc. They are not executed simultaneously. Sometimes, we are getting the following errors: fm_search_defrag(): fm_search_defrag(0x7f212c25a000, 5273984); Free fragment not found!2017-05-26T13:54:20.438736-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f212c25a000, 5273984) called from core: tcp_main.c: tcpconn_new(957), module: core; Free fragment not found!2017-05-26T13:54:20.439101-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:959]: tcpconn_new(): mem. allocation failure2017-05-26T13:54:20.439344-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:3985]: handle_new_connect(): tcpconn_new failed, closing socket I all ready increased this two variables: tcp_conn_wq_max=30000000 tcp_rd_buf_size=5273226 How can I see how much memory xmlrpc is using? Any advice will be appreciated. Thanks in advance! Diego

2 2

Freeing already Freed pointer
by Hamid Hashmi 04 Oct '18

04 Oct '18

I have been receiving CRITICAL core logs since I have updated my service to the following version. Print out of kamailio internals Version: kamailio 5.1.5 (x86_64/linux) d06080 Default config: /usr/local/SIPSer/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/SIPSer/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: d06080 Compiled with: gcc 4.8.5 Compiled on: 14:17:56 Sep 13 2018 Thank you for flying kamailio! I have been starting kamailio with the following command $ /usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamailio/kamailio.cfg -m 1024 -M 512 Sample Critical Logs are below Sep 27 14:40:19 kamailio_eu SIPSer[11289]: 923226363266 923000000001 27872 INVITE haYKxpYM1QR8XTU37WdVbxf4VkEITe7A CRITICAL: <core> [core/mem/q_malloc.c:504]: qm_free(): BUG: freeing already freed pointer (0x7f30c8a4df30), called from tm: t_reply.c: relay_reply(1920), first free tm: h_table.c: free_cell_helper(155) - aborting Sep 27 14:40:19 kamailio_eu SIPSer[11289]: 923226363266 923000000001 27872 INVITE haYKxpYM1QR8XTU37WdVbxf4VkEITe7A CRITICAL: <core> [core/mem/q_malloc.c:504]: qm_free(): BUG: freeing already freed pointer (0x7f30c8a4cc40), called from tm: h_table.c: free_cell_helper(153), first free tm: h_table.c: free_cell_helper(153) - aborting Its also causing service crash. Complete logs are here<https://paste.ee/p/q65DE>. Regards Hamid R. Hashmi

2 2

is_first_hop() fails to detect 'myself' when using advertise
by Sergiu Pojoga 04 Oct '18

04 Oct '18

Hi ppl, If I add a FQDN advertise address to the listen statement, *is_first_hop() <https://kamailio.org/docs/modules/4.3.x/modules/siputils.html#siputils.f.is…>* seems to return FALSE no matter how I play with the listen and alias settings. listen=udp:1.1.1.1:5060 advertise sip.mydomain.net:5060 alias="sip.mydomain.net:5060" This statement doesn't get executed in the *onreply_route* *onreply_route[MANAGE_REPLY] {* *if(is_first_hop())* *set_contact_alias(); # contact alias doesn't get set?* *}* *is_first_hop()* *"The function returns true if the proxy is first hop after the original sender. For incoming SIP replies, it means that top Record-Route URI is 'myself' and source address is not matching it"* *This is the reply.* 2018/10/03 17:03:38.276846 2.2.2.2:62033 -> 1.1.1.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP sip.mydomain.net:5060 ;branch=z9hG4bK49b3.bfbe15.0;received=1.1.1.1 Via: SIP/2.0/UDP 10.10.10.10:5060;rport=5060;branch=z9hG4bK1af96136 *Record-Route*: <sip:sip.mydomain.net;lr=on;did=4cc.734> Contact: <sip:888@10.17.0.35:62033> Suggestions? Thanks.

3 2

What is the typical network setup for kamailio?
by Kevin Olbrich 03 Oct '18

03 Oct '18

Hi! I am working successfully with Kamailio in my lab setup where Kamailio is the SBC for Asterisk. The network layout is looking like this: SIP-Phone <== PUBLIC NET ==> Kamailio (SBC) <== PRIVATE NET ==> Asterisk <== PUBLIC NET ==> Carrier Each public network is reachable from the internet and has a local firewall with IP whitelists. The internal SIP transactions are UDP-only but for external phones I would like to also listen for TCP/TLS. For this layout to work with rtpproxy (before we move on to RTPengine), we have to enable mhomed in Kamailio. We also have some routing issues with packets leaving with the wrong IP via rtpproxy (when call between carrier and external phone needs to be bridged). Most examples show that Asterisk is deployed on the same network as the external interface of Kamailio (-> Asterisk exposed to the public network). In our tests, this works much better but I have great security concerns because this Asterisk instance itself does not need to be reachable from external. How do other users deploy Kamailio in front of Asterisk or similar as SBC to secure internals? There is lot of docs for Kamailio's config but IMHO less for the setup as DMZ (SBC) proxy. Thank you very much. Kind regards Kevin

7 15

Using TLS on a load-balancing cluster behind DNS-SRV
by Kevin Olbrich 03 Oct '18

03 Oct '18

Hi! Which hostname do I need to request for the certificate when the servers are load-balanced using DNS-SRV? Do I need to get the cert for the DNS-SRV subdomain (without _sip._tls) or for the servers, eg. server0{1,2,3}.pbx.example.com ? Thank you! Kevin

3 2

rr module not detecting nat=ws route param
by Jon Bonilla (Manwe) 02 Oct '18

02 Oct '18

Hi all. I'm having problems with an indialog ACK request which is not being routed correctly. I call loose_route and print: $du=sip:MYIP:2443;transport=ws;r2=on;lr=on;did=1d1.7702;nat=ws $route_uri=sip:MYIP;r2=on;lr=on;did=1d1.7702;nat=ws But when I call check_route_param("nat=ws") I get false result. version: kamailio 5.1.4 any hints? cheers, Jon -- https://pekepbx.com https://www.issabel.com/multitenant

3 6

Dialog lifetime for dialogs synced with DMQ
by jenus＠cyberchaos.nl 02 Oct '18

02 Oct '18

Hello, I'm using DMQ to sync dialogs between a active and standby kamailio node. I noticed that when the standby takes over dialog's they do not expire after the lifetime i set on the other node. Is there a way to make sure that calls are terminated after the dialog lifetime expires? I see that the timer is replicated to the other node, but nothing happens after the dialog expires. The event routes aren't executed as well. Thanks, Jan

2 1

Fwd: kamailio with evapi crashing on high volume
by Jayesh Nambiar 02 Oct '18

02 Oct '18

Hello Daniel, Just a quick update on this. After running from the 5.1 branch twice it happened that Server became unresponsive. It never crashed and on forceful shutdown it only generated a core with shutdown routine. Here is what the syslog contained off while the calls were being sent to kamailio but kamailio was not processing any calls: Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first free tm: h_table.c: free_cell_helper(235) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd24a38ea8), called from tm: h_table.c: free_cell_helper(234), first free tm: h_table.c: free_cell_helper(234) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first free tm: h_table.c: free_cell_helper(235) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd24a38ea8), called from tm: h_table.c: free_cell_helper(234), first free tm: h_table.c: free_cell_helper(234) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first free tm: h_table.c: free_cell_helper(235) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd24a38ea8), called from tm: h_table.c: free_cell_helper(234), first free tm: h_table.c: free_cell_helper(234) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first fre And here's the "bt full" of the core file: [New LWP 16003] Cannot access memory at address 0x7fcd64e1a148 Cannot access memory at address 0x7fcd64e1a140 Failed to read a valid object file image from memory. Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/siptrunk.pid -f /usr/local/carrie'. Program terminated with signal SIGABRT, Aborted. #0 0x00007fcd6443f428 in ?? () (gdb) bt full #0 0x00007fcd6443f428 in ?? () No symbol table info available. Backtrace stopped: Cannot access memory at address 0x7ffed40e9e68 Thanks, - Jayesh On Sat, Sep 15, 2018 at 7:09 PM Jayesh Nambiar <jayesh1017(a)gmail.com> wrote: > Hello Daniel, > I have now git cloned from 5.1 branch and restarted kamailio today. Also > an observation; that when one particular customer with around 40-50cps > capacity starts his dialing is when the kamailio kept crashing. I will > observe in the coming week and report of any crashes accordingly. > > Thanks, > > - Jayesh > > On Wed, Sep 12, 2018 at 3:11 AM Jayesh Nambiar <jayesh1017(a)gmail.com> > wrote: > >> Thank you Daniel. I will deploy from 5.1 branch. I will be able to update >> it in next two days. Since this is a production system, I will have to have >> find a proper time slot to get this updated. Will let you know once that is >> done. >> Thanks again. >> >> - Jayesh >> >> On Tue, Sep 11, 2018 at 2:16 PM Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hi Jayesh, >>> >>> I looked over the weekend and seems like a race or a double free. I >>> backported some patches to tm module from master to 5.1 branch (a counter >>> to see if there is more than one free). Would you be able to deploy from >>> branch 5.1 or nightly builds for 5.1? With the next crash I can evaluate >>> (rule out or go for it) if it is double free or not. >>> >>> I haven't written you yet, because I planned to dig more into libev >>> (which is used by evapi), to see if it creates some threads that can mess >>> up with some kamailio globals, although I expect it is event loop in a >>> single thread. Other work related project took a lot of time these days, so >>> not there yet ... >>> Cheers, >>> Daniel >>> >>> On 11.09.18 09:56, Jayesh Nambiar wrote: >>> >>> Hi Daniel, >>> Just a gentle follow up on this. Could you find anything useful? >>> >>> Thanks, >>> >>> - Jayesh >>> >>> >>>>>> On 07.09.18 09:56, Jayesh Nambiar wrote: >>>>>> >>>>>> Hi Daniel, >>>>>> This happens randomly. It is not a testbed and never reproducible on >>>>>> the test server. I can still give you access if you wish to analyse the >>>>>> core files to check whats happening here exactly. >>>>>> It simply feels like on heavy load the evapi socket gives up. Earlier >>>>>> I was running 5.0.2 and in that kamailio would get stuck by not sending any >>>>>> async events on evapi socket. Not even the evapi:connection-closed event >>>>>> gets triggered. After I upgraded to latest stable version (5.1.5), it at >>>>>> least started crashing. Here are two core dumps put onto pastebin. >>>>>> https://pastebin.com/nn6gJapm >>>>>> https://pastebin.com/ph7b8vFH >>>>>> >>>>>> Thanks for all the support, >>>>>> >>>>>> - Jayesh >>>>>> >>>>>> On Fri, Sep 7, 2018 at 1:55 AM Daniel-Constantin Mierla < >>>>>> miconda(a)gmail.com> wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> are you able to reproduce it somehow? Or just happens randomly? >>>>>>> >>>>>>> Is it on a testbed where I could get access to investigate the >>>>>>> corefiles? If not, then I will ask for more details from cores over email, >>>>>>> first would be 'bt full' for both cores and 'p *dead_cell' for the second >>>>>>> one. >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> On 05.09.18 17:47, Jayesh Nambiar wrote: >>>>>>> >>>>>>> Hi Daniel, >>>>>>> Have got these core dumps. Let me know if I should be doing a bt >>>>>>> full. I'll pastebin and send. Thanks, >>>>>>> >>>>>>> Core was generated by `/usr/local/kamailio/sbin/kamailio -P >>>>>>> /var/run/siptrunk.pid -f /usr/local/carrie'. >>>>>>> Program terminated with signal SIGSEGV, Segmentation fault. >>>>>>> #0 0x0000000000505d30 in sip_msg_shm_clone (org_msg=0x7f6ab0d9f618, >>>>>>> sip_msg_len=0x7ffdddb2e8bc, clone_lumps=1) at core/sip_msg_clone.c:491 >>>>>>> 491 LUMP_LIST_LEN(len, org_msg->add_rm); >>>>>>> (gdb) bt >>>>>>> #0 0x0000000000505d30 in sip_msg_shm_clone (org_msg=0x7f6ab0d9f618, >>>>>>> sip_msg_len=0x7ffdddb2e8bc, clone_lumps=1) at core/sip_msg_clone.c:491 >>>>>>> #1 0x00007f6af0bdf68d in fake_req (shmem_msg=0x7f6ab0d9f618, >>>>>>> extra_flags=0, uac=0x7f6ab1738980, len=0x7ffdddb2e8bc) at t_reply.c:854 >>>>>>> #2 0x00007f6af0c3aa27 in t_continue_helper (hash_index=58039, >>>>>>> label=413633661, rtact=0x7f6af10500f0, cbname=0x0, cbparam=0x0) at >>>>>>> t_suspend.c:293 >>>>>>> #3 0x00007f6af0c3eed4 in t_continue (hash_index=58039, >>>>>>> label=413633661, route=0x7f6af10500f0) at t_suspend.c:583 >>>>>>> #4 0x00007f6aae4dd010 in w_t_continue (msg=0x7ffdddb2fa60, >>>>>>> idx=0x7f6af1098e90 "8\306\t\361j\177", lbl=0x7f6af1098ff0 >>>>>>> "\240\264\t\361j\177", rtn=0x7f6af1099150 "0\275\t\361j\177") at >>>>>>> tmx_mod.c:760 >>>>>>> #5 0x000000000045b477 in do_action (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af109ab38, msg=0x7ffdddb2fa60) at core/action.c:1085 >>>>>>> #6 0x0000000000467fd5 in run_actions (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af1096630, msg=0x7ffdddb2fa60) at core/action.c:1565 >>>>>>> #7 0x000000000045b234 in do_action (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af10a0f80, msg=0x7ffdddb2fa60) at core/action.c:1058 >>>>>>> #8 0x0000000000467fd5 in run_actions (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af10a0f80, msg=0x7ffdddb2fa60) at core/action.c:1565 >>>>>>> #9 0x0000000000468797 in run_top_route (a=0x7f6af10a0f80, >>>>>>> msg=0x7ffdddb2fa60, c=0x0) at core/action.c:1654 >>>>>>> #10 0x00007f6aabe79370 in evapi_run_cfg_route (evenv=0x7ffdddb30250, >>>>>>> rt=3, rtname=0x7f6aac08cb18 <_evapi_rts+56>) at evapi_dispatch.c:161 >>>>>>> #11 0x00007f6aabe7f271 in evapi_recv_client (loop=0x7f6aabe698e0, >>>>>>> watcher=0x27af5e0, revents=1) at evapi_dispatch.c:467 >>>>>>> #12 0x00007f6aabc5fd73 in ev_invoke_pending () from >>>>>>> /usr/lib/x86_64-linux-gnu/libev.so.4 >>>>>>> #13 0x00007f6aabc633de in ev_run () from >>>>>>> /usr/lib/x86_64-linux-gnu/libev.so.4 >>>>>>> #14 0x00007f6aabe7867c in ev_loop (loop=0x7f6aabe698e0, flags=0) at >>>>>>> /usr/include/ev.h:835 >>>>>>> #15 0x00007f6aabe83fc6 in evapi_run_dispatcher (laddr=0x7f6af0f72300 >>>>>>> "127.0.0.1", lport=8060) at evapi_dispatch.c:705 >>>>>>> #16 0x00007f6aabe6e262 in child_init (rank=0) at evapi_mod.c:213 >>>>>>> #17 0x0000000000542cad in init_mod_child (m=0x7f6af0f71b70, rank=0) >>>>>>> at core/sr_module.c:943 >>>>>>> #18 0x0000000000542971 in init_mod_child (m=0x7f6af0f72968, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #19 0x0000000000542971 in init_mod_child (m=0x7f6af0f73d38, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #20 0x0000000000542971 in init_mod_child (m=0x7f6af0f74670, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #21 0x0000000000542971 in init_mod_child (m=0x7f6af0f76708, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #22 0x0000000000542971 in init_mod_child (m=0x7f6af0f76c08, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #23 0x0000000000542971 in init_mod_child (m=0x7f6af0f770d0, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #24 0x0000000000542971 in init_mod_child (m=0x7f6af0f77cf0, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #25 0x0000000000542971 in init_mod_child (m=0x7f6af0f78808, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #26 0x0000000000542971 in init_mod_child (m=0x7f6af0f78bd8, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #27 0x0000000000542971 in init_mod_child (m=0x7f6af0f794c8, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #28 0x0000000000542971 in init_mod_child (m=0x7f6af0f79920, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #29 0x0000000000542971 in init_mod_child (m=0x7f6af0f7a330, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #30 0x0000000000542971 in init_mod_child (m=0x7f6af0f7afd0, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #31 0x0000000000542971 in init_mod_child (m=0x7f6af0f7bc80, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #32 0x000000000054303d in init_child (rank=0) at core/sr_module.c:970 >>>>>>> #33 0x0000000000425399 in main_loop () at main.c:1701 >>>>>>> #34 0x000000000042bd5c in main (argc=13, argv=0x7ffdddb31088) at >>>>>>> main.c:2638 >>>>>>> >>>>>>> And this: >>>>>>> [New LWP 15804] >>>>>>> [Thread debugging using libthread_db enabled] >>>>>>> Using host libthread_db library >>>>>>> "/lib/x86_64-linux-gnu/libthread_db.so.1". >>>>>>> Core was generated by `/usr/local/kamailio/sbin/kamailio -P >>>>>>> /var/run/siptrunk.pid -f /usr/local/carrie'. >>>>>>> Program terminated with signal SIGSEGV, Segmentation fault. >>>>>>> #0 0x00007f6af0b90b70 in free_cell_helper >>>>>>> (dead_cell=0x7f6ab0a6baa8, silent=0, fname=0x7f6af0c8f630 "timer.c", >>>>>>> fline=654) at h_table.c:230 >>>>>>> 230 foo = tt->next; >>>>>>> (gdb) bt >>>>>>> #0 0x00007f6af0b90b70 in free_cell_helper >>>>>>> (dead_cell=0x7f6ab0a6baa8, silent=0, fname=0x7f6af0c8f630 "timer.c", >>>>>>> fline=654) at h_table.c:230 >>>>>>> #1 0x00007f6af0c24409 in wait_handler (ti=932640643, >>>>>>> wait_tl=0x7f6ab0a6bb28, data=0x7f6ab0a6baa8) at timer.c:654 >>>>>>> #2 0x00000000004bb445 in timer_list_expire (t=932640643, >>>>>>> h=0x7f6ab03ad158, slow_l=0x7f6ab03ae480, slow_mark=271) at core/timer.c:874 >>>>>>> #3 0x00000000004bb8ab in timer_handler () at core/timer.c:939 >>>>>>> #4 0x00000000004bbd30 in timer_main () at core/timer.c:978 >>>>>>> #5 0x00000000004250f9 in main_loop () at main.c:1691 >>>>>>> #6 0x000000000042bd5c in main (argc=13, argv=0x7ffdddb31088) at >>>>>>> main.c:2638 >>>>>>> >>>>>>> On Wed, Sep 5, 2018 at 3:13 PM Daniel-Constantin Mierla < >>>>>>> miconda(a)gmail.com> wrote: >>>>>>> >>>>>>>> Hello, >>>>>>>> >>>>>>>> the backtrace doesn't show any hint about kamailio, only from >>>>>>>> closelog() up. >>>>>>>> >>>>>>>> It may be the core generated by shutdown procedure, have you >>>>>>>> enabled one core file per pid/process? If not, do it and reproduce the >>>>>>>> issue again, you may get two core files, one being the runtime issue and >>>>>>>> the other one from shutdown procedure, which likely is an effect of the >>>>>>>> other one. The one from the runtime is more relevant. >>>>>>>> >>>>>>>> Cheers, >>>>>>>> Daniel >>>>>>>> >>>>>>>> On 05.09.18 10:09, Jayesh Nambiar wrote: >>>>>>>> >>>>>>>> Hello, >>>>>>>> I'm using kamailio 5.1.5 with evapi. I have a node.js connecting >>>>>>>> with kamailio evapi to which I send events and also consume events based on >>>>>>>> which I do the routing. I have 8 evapi workers defined in the config. >>>>>>>> The problem is that kamailio randomly crashes on high load. I'm >>>>>>>> assuming that it is related to the evapi module as rest of the config is >>>>>>>> pretty straight forward. I could get a core file and here's the core dump: >>>>>>>> [New LWP 14042] >>>>>>>> [Thread debugging using libthread_db enabled] >>>>>>>> Using host libthread_db library >>>>>>>> “/lib/x86_64-linux-gnu/libthread_db.so.1”. >>>>>>>> Core was generated by `/usr/local/kamailio/sbin/kamailio -P >>>>>>>> /var/run/siptrunk.pid -f /usr/local/carrie’. >>>>>>>> Program terminated with signal SIGABRT, Aborted. >>>>>>>> #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at >>>>>>>> ../sysdeps/unix/sysv/linux/raise.c:54 >>>>>>>> 54 ../sysdeps/unix/sysv/linux/raise.c: No such file or directory. >>>>>>>> (gdb) bt >>>>>>>> #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at >>>>>>>> ../sysdeps/unix/sysv/linux/raise.c:54 >>>>>>>> #1 0x00007f999528502a in __GI_abort () at abort.c:89 >>>>>>>> #2 0x000000000041a029 in sig_alarm_abort (signo=14) at main.c:646 >>>>>>>> #3 <signal handler called> >>>>>>>> #4 0x00007f999534f497 in __libc_cleanup_routine (f=<optimized >>>>>>>> out>) at ../sysdeps/nptl/libc-lockP.h:291 >>>>>>>> #5 closelog () at ../misc/syslog.c:415 >>>>>>>> #6 0x0000000000000000 in ?? () >>>>>>>> >>>>>>>> Any help in this regards is would allow me to identify the reason >>>>>>>> of the crash. Thanks for the support. >>>>>>>> >>>>>>>> - Jayesh >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>>>>> Kamailio World Conference -- www.kamailioworld.com >>>>>>>> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com >>>>>>>> >>>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>>>> Kamailio World Conference -- www.kamailioworld.com >>>>>>> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com >>>>>>> >>>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>>> Kamailio World Conference -- www.kamailioworld.com >>>>>> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com >>>>>> >>>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>> Kamailio World Conference -- www.kamailioworld.com >>>>> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com >>>>> >>>>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Kamailio World Conference -- www.kamailioworld.com >>> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com >>> >>>

2 2

[DMQ] dmq_send_message function usage.
by Denys Pozniak 02 Oct '18

02 Oct '18

Hello! I have set of Kamailios which distributed location via dmq_usrloc. Could I construct synthetic DMQ message from separate Kamailio node to this cluster to get location info for the specific user? If yes, please explain how to do it. -- BR, Denys Pozniak

3 3

Forcing outbound transport to TCP
by Ivan Ribakov 02 Oct '18

02 Oct '18

I’m new to Kamailio and currently evaluating it’s capabilities. One of the things I would like to be able to do is to change outbound message transport based on some dynamic logic. Kamailio setup: I’m running on a setup from this guide - http://kamailio.org/docs/tutorials/5.1.x/kamailio-install-guide-git/ <http://kamailio.org/docs/tutorials/5.1.x/kamailio-install-guide-git/> and using default configuration file. In a basic scenario of one2one call, I was able to make Kamailio to forward INVITE to callee over TCP by issuing a REGISTER request with “transport=tcp” parameter. Although it worked, as far as I understand that means all communication with the callee will be happening over TCP. I would like to be able to change transport dynamically so looking for a way to do that from config file. I tried modifying $ru/$du in the following manner, but it didn’t seem to work. # default config content if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting # added to try to force TCP transport $du = $du + ";transport=tcp"; } Any pointers as to how can I modify the outbound message transport? Do I need to use some extra modules or can it be achieved with a basic setup? Thanks in advance, Ivan

4 5

Registrar Module
by Dan Quinney 01 Oct '18

01 Oct '18

Hi, So, route[REGISTRAR_1] will, when the max_contacts is reached, reply to the endpoint with a 503 error containing a "P-Registrar-Error: Too many registered contacts" header - this seems wrong to me so I've tried to work around it with route[REGISTRAR_2] which returns the right response, but, also writes "INFO: registrar [save.c:615]: test_max_contacts(): too many contacts for AOR" to the log file, which is messy. Does anyone have any pointers for doing this better? Thanks, Dan route[REGISTRAR_1] { if ( is_method("REGISTER") ) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if ($avp(maximum_contacts) != $null) { $xavp(reg=>max_contacts) = $avp(maximum_contacts); } save("location"); } return; } route[REGISTRAR_2] { if ( is_method("REGISTER") ) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if ($avp(maximum_contacts) != $null) { $xavp(reg=>max_contacts) = $avp(maximum_contacts); } save("location", "0x02"); switch ($retcode) { case -1: sl_send_reply("500", "Server Internal Error"); exit; case -2: sl_send_reply("403", "Forbidden"); exit; }; sl_send_reply("200", "OK"); } return; }

2 1

RTPEngine Question
by Wilkins, Steve 01 Oct '18

01 Oct '18

Hello All, I have an issue where Kamailio-RTPEngine-Asterisk calls work good when a softphone UAC on an IOS phone makes an Inbound call to a WebRTC client. However, if the softphone UAC is on Windows, it does not work (No Audio/Video). I noticed in the Wireshark trace that when using IOS I see STUN packets, but when using Windows I see no STUN packets. I believe this may be my issue, but I am not sure why the IOS would create STUN packets but when using Windows I get no STUN packets. Does anyone have any thoughts on what could be causing this? Thank you,

1 0

Re: [SR-Users] Problem with replicating dialogs using DMQ
by Charles Chance 01 Oct '18

01 Oct '18

Hi Jan, Setting multi_notify parameter should fix. Cheers, Charles On Mon, 1 Oct 2018 at 08:26, <jenus(a)cyberchaos.nl> wrote: > Charles, > > I have the follwing parameters configured: > > # ----- dmq params ----- > modparam("dmq", "server_address", "sip:10.10.10.1:5060") > modparam("dmq", "num_workers", 4) > modparam("dmq", "ping_interval", 90) > modparam("dmq", "notification_address", > "sip:sip04.voip.domain.test:5060") > > And the output of dmq.list_nodes is: > > { > host: 10.10.10.2 > port: 5060 > resolved_ip: 10.10.10.2 > status: active > last_notification: 0 > local: 0 > } > { > host: sip03.voip.domain.test > port: 5060 > resolved_ip: 10.10.10.1 > status: active > last_notification: 0 > local: 0 > } > { > host: sip04.voip.domain.test > port: 5060 > resolved_ip: 10.10.10.2 > status: active > last_notification: 0 > local: 0 > } > { > host: 10.10.10.1 > port: 5060 > resolved_ip: 10.10.10.1 > status: active > last_notification: 0 > local: 1 > } > > > As you can see the nodes are listed twice, one time with the ip and one > time with the hostname. If i change the notification_address to a ip > address the duplicate entry's are gone. > > Thanks, > > Jan > > Charles Chance schreef op 2018-09-28 20:54: > > Hello, > > > > Can you share your dmq config (modparams) and the output of > > dmq.list_nodes? > > > > Cheers, > > > > Charles > > > > On Thu, 27 Sep 2018 at 12:51, Jan Hazenberg <jenus321(a)gmail.com> > > wrote: > > > >> In the output of "kamcmd dmq.list_nodes" i see nodes twice. One with > >> ip address of the host and one with the hostname of the host. > >> > >> Jan > >> > >> On Thu, Sep 27, 2018 at 12:47 PM Jan Hazenberg <jenus321(a)gmail.com> > >> wrote: > >> > >> Daniel, > >> > >> It indeed sends duplicate DMQ messages, one to the ip of the other > >> node and one to the hostname of the other node. > >> > >> Unknown request: KDMQ sip:dialog@sip04.voip.domain.acc:5060 | > >> (application/json) > >> Unknown request: KDMQ sip:dialog@10.10.10.12:5060 [5] | > >> (application/json) > >> > >> Is there any way to disable this? > >> > >> Thanks, > >> > >> Jan > >> > >> On Thu, Sep 27, 2018 at 11:50 AM Daniel-Constantin Mierla > >> <miconda(a)gmail.com> wrote: > >> > >> Hello, > >> > >> the log message says that the dialog was already added to the timer. > >> Do you see retransmissions of the DMQ requests on the network with > >> ngrep/wireshark/...? > >> Cheers, > >> Daniel > >> > >> On 27.09.18 10:40, Jan Hazenberg wrote: > >> > >> Hello, > >> > >> I'm trying to replicate active dialogs using the DMQ module. I see > >> the dialogs on the other nodes but it generates a error message in > >> the logs on all the nodes: > >> > >> Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog > >> [dlg_timer.c:130]: insert_dlg_timer(): Trying to insert a bogus dlg > >> tl=0x7efdca1bebf8 tl->next=0x7efdca14cb80 tl->prev=0x7efdca14cb80 > >> Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog > >> [dlg_dmq.c:309]: dlg_dmq_handle_msg(): Unable to insert dlg timer > >> 0x7efdca1beb98 [2643:3725] > >> > >> Even with the timeout_avp option disabled i still see the error > >> messages. I tested this on kamailio 5.0.3 and 5.1.3 with the same > >> results. > >> > >> With debug enabled i see the correct data is replicated to the other > >> nodes: > >> > >> dlg_dmq_replicate_action(): sending serialized data > >> > > > {"action":2,"h_entry":1328,"h_id":409,"state":4,"start_ts":1538036606,"lifetime":7200,"tag1":"1635359980","tag2":"as2d80799c","cseq1":"21","cseq2":"0","route_set1":"","route_set2":"","contact1":" > sip:sipuser@10.10.10.10:1057;alias=10.10.10.10~1055~1","contact2":" > sip:+31101234567@10.10.10.10:5060 > >> [6]"} > >> > >> "lifetime":7200 is in the serialized data, and when i check on the > >> other node with "kamcmd dlg.list" i see that the lifetime data is > >> not updated. > >> > >> Thanks, > >> > >> Jan > >> > >> _______________________________________________ > >> Kamailio (SER) - Users Mailing List > >> sr-users(a)lists.kamailio.org > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >> > >> -- > >> Daniel-Constantin Mierla -- www.asipto.com [1] > >> www.twitter.com/miconda [2] -- www.linkedin.com/in/miconda [3] > >> Kamailio World Conference -- www.kamailioworld.com [4] > >> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- > >> www.asipto.com [1] > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > > > > CHARLES CHANCE > > Managing Director > > > > t. 0330 120 1200 m. 07932 063 891 > > Sipcentric Ltd. Company registered in England & Wales no. 7365592. > > Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt > > Street, Birmingham Science Park, Birmingham B7 4BB. > > > > Links: > > ------ > > [1] http://www.asipto.com > > [2] http://www.twitter.com/miconda > > [3] http://www.linkedin.com/in/miconda > > [4] http://www.kamailioworld.com > > [5] http://sip:dialog@10.10.10.12:5060 > > [6] http://sip:+31101234567@10.10.10.10:5060 > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- *Charles Chance* Managing Director t. 0330 120 1200 m. 07932 063 891 -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

2 1

Problem with replicating dialogs using DMQ
by Jan Hazenberg 01 Oct '18

01 Oct '18

Hello, I'm trying to replicate active dialogs using the DMQ module. I see the dialogs on the other nodes but it generates a error message in the logs on all the nodes: Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_timer.c:130]: insert_dlg_timer(): Trying to insert a bogus dlg tl=0x7efdca1bebf8 tl->next=0x7efdca14cb80 tl->prev=0x7efdca14cb80 Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_dmq.c:309]: dlg_dmq_handle_msg(): Unable to insert dlg timer 0x7efdca1beb98 [2643:3725] Even with the timeout_avp option disabled i still see the error messages. I tested this on kamailio 5.0.3 and 5.1.3 with the same results. With debug enabled i see the correct data is replicated to the other nodes: dlg_dmq_replicate_action(): sending serialized data {"action":2,"h_entry":1328,"h_id":409,"state":4,"start_ts":1538036606,"lifetime":7200,"tag1":"1635359980","tag2":"as2d80799c","cseq1":"21","cseq2":"0","route_set1":"","route_set2":"","contact1":"sip:sipuser@10.10.10.10:1057 ;alias=10.10.10.10~1055~1","contact2":"sip:+31101234567@10.10.10.10:5060"} "lifetime":7200 is in the serialized data, and when i check on the other node with "kamcmd dlg.list" i see that the lifetime data is not updated. Thanks, Jan

4 5

registrar lookup with different URI
by Maarten Ureel 30 Sep '18

30 Sep '18

Hello I want the registar module to lookup a user with a different 'name' as the RURI. We have the extension & domain in variables, but the RURI is ext@ip-of-server, e.g. 200(a)1.1.1.1<mailto:200@1.1.1.1>, while we want to perform the lookup for 200(a)somedomain.tld<mailto:200@somedomain.tld>. We have somedomain.tld in a header, so we have it in a variable. However, if we perform the lookup like this (using Python module): lookup_result = KSR.registrar.lookup('location') We get a negative result, and the debug logs show that it tries to locate 200(a)1.1.1.1<mailto:200@1.1.1.1> - which makes sense. So according to docs you can specify an URI, however when I do this: lookup_result = KSR.registrar.lookup('location' '200(a)somedomain.tld') lookup_result is "False" and the function is not executed, according to docs because it cannot obtain the parameters for this function. How can we force the registrar module to lookup a different user?

1 0

Dialog timout reset when receiving a re-INVITE
by jenus＠cyberchaos.nl 30 Sep '18

30 Sep '18

Hello, I have a issue with setting the dialog timeout for a call. It works fine until i receive a re-INVITE. The re-INVITE seems to reset the dialog timeout back to the default value configured in the module parameters (7200 in my case). I use "dlg_set_timeout("900");" to set the timeout to 900 seconds. When i receive a re-INVITE i see the following in the debug logs: DEBUG: dialog [dlg_handlers.c:647]: get_dlg_timeout(): invalid AVP value, using default timeout This happens even with the "timeout_avp" module parameter disabled. I'm running kamailio 5.1.3. Thanks, Jan

3 2

Restoring htable shared variables via dmq after Kamailio restart
by Denys Pozniak 28 Sep '18

28 Sep '18

Hello! I have 2 nodes that replicates htable variables via DMQ. How can I recover them after one node restarting? -- BR, Denys Pozniak

3 2

kamailio rtp connection
by Pieter Muller 28 Sep '18

28 Sep '18

Good day all I hope Im on the right place to ask the question. Im a newbie to kamailio and have a little problem understanding how to set kamailio correctly. Scenario: VoIP Supplier (IP:161.161.252.20{Private Net}) à Connect via routing to Kamailio Server(155.155.16.2 eno2{private Net}),Kamailio Server(187.221.197.252 eno1{public facing})àconnect to PBX(public IP set in dispatcher) à connect to RTP server(Public IP) Now the problem is that if you call in Number(a)161.161.252.20 <mailto:Number@161.161.252.20> is send to kamailio, kamailio then sends it down to dispatcher IP and that PBX responds directly back via rtp server to 161.161.252.20. The 161.161.252.20 IP is private and cannot be reached from the cloud and vice versa . How do I get kamailio to be the middle man between 161.161.252.20 and RTP server(all traffic must go via the kamailio server) Is there specific modules I must load. Im running Kamailio 4.4 Any help would be muchly appreciated. Regards Pieter

2 3

Important: new module exports interface
by Daniel-Constantin Mierla 28 Sep '18

28 Sep '18

Hello, as a result of collaborative work at Kamailio Developers Meeting, we succeeded to merge the two existing module exports interface (one for kamailio modules and the other one from ser modules) in a single one. All public modules were updated, but if you have any private module then you have to update as well in order to get it compiled with the latest master. Just look at one of the modules (e.g., sl is a good option) and all the fields in mod exports structure have comments with their meaning. In short: we removed unused fields for statistics, mi commands and extra-processes (from kamailio old interface) and oncancel (from ser old interface), kept the rpc exports from ser interface and pv exports and dlopen flags from kamailio interface. The other fields were common in both interfaces, but be aware that order was also changed. If you have a module implementing the old ser interface, you also need to add the free_fixup_field in functions exports tructure -- you can just set it to 0. With this we have now a single mod exports interface in all modules, the core is also cleaner as we got rid of the various compatibility layers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 0

INFO: maintenance work on deb.kamailio.org
by Daniel-Constantin Mierla 28 Sep '18

28 Sep '18

Hello, we will do some maintenance work on deb.kamailio.org server these days in order to move to a better infrastructure, so it may not be available for a while -- the DNS needs to be updated as well, this also can affect you if your router/provider does caching. Once we finish with the migration, we will send updates. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 1

Re: [SR-Users] No dispatcher failover
by Gertjan Wolzak 28 Sep '18

28 Sep '18

Alex, Thank you for the feedback, just the info I needed. Failover is working now just as it should. Now I just have to tweak the failover timing. Rgds, Gertjan Wolzak

1 0

Cannot handle Cancel Request properly
by Aqs Younas 28 Sep '18

28 Sep '18

Greetings list, I am getting hard time to make kamailio perfectly handle cancels request. I am using http_async_client and MSILO to handle push notifications. Case 1. When B party record exists in location but it is off-line and I send cancel *instantly* from Party A while kamailio is sending push to wake B Party up. modparam("tm", "failure_reply_mode", 3) modparam("tm", "cancel_b_method", 1) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) You can see in below picture 407 being generated after some time. [image: default.PNG] What could be the reason of this problem? any pointer is much appreciated. Br, Aqs.

2 1

replay of tls delayed by 30 secs.
by Andrew Chen 28 Sep '18

28 Sep '18

Hi Kam community, Facing an urgent issue here and not sure how to troubleshoot it further. So our engineers been getting reports of unknown SIP timeouts and they suspect it's something on the Kamailio side. I've logged every transaction from receiving the message to done relaying it as shown below: Sep 24 18:38:13 sjoprodfnsg01 /usr/local/sbin/kamailio[42608]: INFO: <script>: [1589Fz-ses1-41d9a88aR70125425(a)xxx.xxx.xxx.xxx][4534123][CSeq 7] NG-FuzeMedia-Server/42ae58c/mediahub Relaying [UPDATE] from (xxx.xxx.xxx.xxx ) to ( xxx.xxx.xxx.xxx ) via (tls) Sep 24 18:38:13 sjoprodfnsg01 /usr/local/sbin/kamailio[42608]: INFO: <script>: [1589Fz-ses1-41d9a88aR70125425@ xxx.xxx.xxx.xxx ][4534123][CSeq 7] NG-FuzeMedia-Server/42ae58c/mediahub Done relay [UPDATE] from (xxx.xxx.xxx.xxx ) to ( xxx.xxx.xxx.xxx ) via (tls) When I pulled the 443 SIP trace from my tcpdump session, it shows the application data left the machine at 18:44 as shown here: [image: image.png] Note: This packet is encrypted and can't see the actual SIP message. This matches what was received on the client logs: 2018-09-24T18:38:44.032Z [40927] [INF] fuze.sip SipParser.cpp:549 LogSipMessage: xxx.xxx.xxx.xxx:443 - 4292B (local time 09/24 11:38:44) -------------------- RECEIVE: UPDATE sips:Luke_Surazski@ xxx.xxx.xxx.xxx:50795;id=37262838;inst=52241254 SIP/2.0 Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:443;branch=z9hG4bK15f.5052c318e1766a0a9405f10708f43188.0;i=acbb7 Via: SIP/2.0/TCP xxx.xxx.xxx.xxx :38342;rport=38342;branch=z9hG4bK-55f913-t9-I214262897 What would cause such delays? I'm running Kamailio 4.1.5 Thanks. -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 9

subst_hf to rewrite a contact header
by Ben Hood 27 Sep '18

27 Sep '18

Hi, I’m wondering how to use subst_hf() properly. I’m trying to debug a broken flow by manually rewriting the contact header from Contact: <sip:b73c6f29-0101-4802-afcd-efb63f1e6d8f@10.10.10.12:5090;transport=udp> to Contact: <sip:2018092417381900003@81.x.x.x> by using $var(ctct) = "<sip:2018092417381900003@81.x.x.x>"; subst_hf("Contact", “/\<.+\>/$var(ctct)\r\n/", "a”); but the result is Contact: <sip:2018092417381900003@81.x.x.x> sip:b73c6f29-0101-4802-afcd-efb63f1e6d8f@10.10.10.12:5090;transport=udp Testing the regex with an external tool appears to match all of the original header, so I’m wondering why subst_hf doesn’t appear to rewrite the entire header. Is there something I’m doing wrong? TIA, Ben

3 7

Documentation of Kamailio.cfg
by Prabhat Kumar 27 Sep '18

27 Sep '18

Hi everyone, I there any suitable documentation for understating kamailio configuration file. Regards, Prabhat Kumar Sr. Software Developer [image: phone] +91 92129 92129 Ext: 89 [image: moible] +91 80831 10443 [image: Facebook] <https://app.mailtag.io/link-event-v2?mt__id=379f500e-c250-11e8-a3de-0625822…>[image: Twitter] <https://app.mailtag.io/link-event-v2?mt__id=379f5248-c250-11e8-9d85-0625822…>[image: Twitter] <https://app.mailtag.io/link-event-v2?mt__id=379f5392-c250-11e8-b97b-0625822…>[image: Linkedin] <https://app.mailtag.io/link-event-v2?mt__id=379f54be-c250-11e8-8e23-0625822…>[image: Google+] <https://app.mailtag.io/link-event-v2?mt__id=379f55f4-c250-11e8-affd-0625822…> [image: Logo] <https://app.mailtag.io/link-event-v2?mt__id=379f570c-c250-11e8-a472-0625822…>

2 1

Kamailio - Asterisk integration
by denislav.tsonev＠musala.com 27 Sep '18

27 Sep '18

Hello guys, I'm following this guide: https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And I have the following schema: Phone A -----> Kamailio -----> Asterisk -----> Phone B (10.0.8.249) (10.0.6.123) (10.0.3.144) (10.0.13.23) I think the flow it is like (because when UAC register to Asterisk, the contact points to Kamailio IP): 1. Phone A --> Kamailio --> Asterisk 2. Asterisk --> Kamailio --> Phone B In attachment is the pcap of the call, kamailio config. The problem is that I never receive ACK after 200 OK and call drops after 30 seconds? Kamailio version 5.1 and Asterisk version 15.1 Phones are grandstreams and also tested with microsip soft client and same results... Like "Phone A" never sends ACK back I tried different approaches to change contact header in 200 OK, because is Asterisk IP. Disable NAT handling. Different "nat_uac_test" flags, but still without success :( Also tried with this NAT detection and manage routes (again without any success) - source https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } Thanks in advance Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

1 0

Problem with replicating dialogs using DMQ
by jenus＠cyberchaos.nl 27 Sep '18

27 Sep '18

Hello, I'm trying to replicate active dialogs using the DMQ module. I see the dialogs on the other nodes but it generates a error message in the logs on all the nodes: Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_timer.c:130]: insert_dlg_timer(): Trying to insert a bogus dlg tl=0x7efdca1bebf8 tl->next=0x7efdca14cb80 tl->prev=0x7efdca14cb80 Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_dmq.c:309]: dlg_dmq_handle_msg(): Unable to insert dlg timer 0x7efdca1beb98 [2643:3725] Even with the timeout_avp option disabled i still see the error messages. I tested this on kamailio 5.0.3 and 5.1.3 with the same results. With debug enabled i see the correct data is replicated to the other nodes: dlg_dmq_replicate_action(): sending serialized data {"action":2,"h_entry":1328,"h_id":409,"state":4,"start_ts":1538036606,"lifetime":7200,"tag1":"1635359980","tag2":"as2d80799c","cseq1":"21","cseq2":"0","route_set1":"","route_set2":"","contact1":"sip:sipuser@10.10.10.10:1057;alias=10.10.10.10~1055~1","contact2":"sip:+31101234567@10.10.10.10:5060"} "lifetime":7200 is in the serialized data, and when i check on the other node with "kamcmd dlg.list" i see that the lifetime data is not updated. Thanks, Jan

1 0

create domain "on the fly" in registrar modukle
by Maarten Ureel 27 Sep '18

27 Sep '18

I'm building a setup with FreeSWITCH and Kamailio 5.2 but I'm having troubles in handling the registrations for multiple domains. So far I have in my config these related settings: # Disable database completely modparam("usrloc", "db_mode", 0) # Take domain into account when saving modparam("usrloc", "use_domain", 1) # Preload location table modparam("usrloc", "preload", "location") I'm using the app_python module; when performing the registrar save like this: KSR.registrar.save(KSR.pv.get('$fd'), 1) It fails, the debug shows that it cannot find the domain that the SIP user has here. When I use: KSR.registrar.save('location', 1) The phone registers just fine; however I'm bothered by the fact that the documentation states that you should specify the domain as a parameter. Is it OK if everything is in one 'location' domain? How can I let Kamailio know what domains exist (where does Kamailio get this information from)? We have a unique domain for every customer, like g123.ctx.ourcompany.tld. Kind regards Maarten Ureel

2 1

Adding new rtpproxy instance without restarting Kamailio
by Pintu Lohar 27 Sep '18

27 Sep '18

Hi All, I have a very basic question. Is there any way to add new rtpproxy instance without restarting Kamailio? As per my understanding, we can reload the dispatcher table when Kamailio server is running with kamctl dispatcher reload but do we have any such for rtpproxy list also from db when rtpproxy table ? Below is my rtpproxy related config, load from DB , #Once RTPProxy was found unreachable and marked as disabled, #the rtpproxy module will not attempt to establish communication to RTPProxy for rtpproxy_disable_tout seconds. modparam("rtpproxy", "rtpproxy_disable_tout", 120) #only for local host. modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2") modparam("rtpproxy", "db_url", "mysql://kamailio:xxxxxxxx@10.xx.xx.xx /kamailio") modparam("rtpproxy", "table_name", "rtpproxy") #enable marker for sdp modification by rtpproxy modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n") my current instance : ./kamcmd rtpproxy.list { setid: 0 url: udp:xxx.xx.xxx.xxx:7722 index: 2 disabled: 0 weight: 1 recheck: 0 } Db +----+-------+--------------------------+-------+--------+-------------+ | id | setid | url | flags | weight | description | +----+-------+--------------------------+-------+--------+-------------+ | 1 | 0 | udp:XXX.XXX.XXX.XXX:7722 | 0 | 1 | | +----+-------+--------------------------+-------+--------+-------------+ I already explored ./kamcmd rtpproxy.enable command , but it seems that this command is useful for re-enabling or disabling existing rtpproxy instance . My purpose is to add new rtpproxy instance when Kamailio is running. Thanks in advance. Any suggestion would very helpful. Thanks & Regards Pintu

1 0

KAMAILIO 5.1 : How to enable and subscribe to all of the SIP events and kamailio server through tcp, json
by yboujraf＠by-systems.be 26 Sep '18

26 Sep '18

Dear All, I am currently working to control from third party system Kamailio I have there are several way to interact with Kamailio but not sure I could make what I am expecting My question is : * I would like to get from my third party system subscribe to all of the sip events and status of the kamailio server My best way is TCP, JSON, scrambled if possible It will be great if anybody could route me to the correct direction. Best Regards, Youssef --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

1 0

KAMAILIO 5.1 - how to make a full redundancy of two kamailio Proxy server ?
by yboujraf＠by-systems.be 26 Sep '18

26 Sep '18

Dear All, I am currently thinking to use TWO (2) Kamailio Proxy server in redundancy. The system is : * I have one (1) public address * Two (2) Kamailio Servers with two different IP address My questions are : * What I need to add in front of the two Kamailio servers to be sure sip phone will be connected through the public ip address to one of the kamailio server at time. I have seen we could enable "cluster module" to sync the dabase of the two (2) kamailio server at the same time but not for the request. * Script to add to be sure the kamailio servers are correctly configured. Best Regards, Youssef --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

1 0

kamailio not forwarding 407
by David Villasmil 26 Sep '18

26 Sep '18

Hello guys, I have an almost default config, the box behind kamailio sends out the register, it is forwarded to the provider, which replies with 401, this gets forwarded back to the box which sends out the REGISTER with credentials and all works well. BUT when calling out, INVITE is sent to kamailio which is properly forwarded to the provider. The provider replies with a 407 and this 407 is NOT forwarded back to the box. Any ideas why? I can't figure it out! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Dispatcher's Probing
by Duarte Rocha 26 Sep '18

26 Sep '18

Hey, I've been using the Dispatcher module with great results. But now, i want to be able to have some gateways to belong to more than one group set. To achieve that, i'm duplicating the Gateway in the Dispatcher's table and give it a different group set. However, if i do this, Dispatcher will do repeated probing to this gateways. Is there any way to avoid this or to achieve the same result in a different way? Best Regards

1 0

No dispatcher failover
by Gertjan Wolzak 26 Sep '18

26 Sep '18

Hello Kamailions, I want to send traffic to an Asterisk box but when that box fails traffic should be routed to the second Asterisk box. The dispatcher params are as follows: # ----- Dispatcher params ------ modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") modparam("dispatcher", "ds_inactive_threshold", 4) modparam("dispatcher", "ds_ping_latency_stats", 1) modparam("dispatcher", "ds_probing_threshold", 1) My dispatcher.list is: #DBBS PBX # SBC's 1 sip:ip_address_sbc:5060 0 0 weight=100 #PBX 2 sip:ip_pbx_01:5060 0 10 weight=100 2 sip:ip_pbx_02:5060 0 5 weight=100 Verry simple request route: request_route { if (is_method("OPTIONS")) { sl_send_reply("200", "OK"); exit; } #Call from SBC to PBX #Get gateway from dispatcher list if($si=="ip_address_sbc1" |$si=="ip_address_sbc2" ) { ds_select_dst("2", "8"); $var(new_uri) = "sip:" + $tU + "@" + $dd + ":5060"; $ru=($var(new_uri)); route(RELAY); exit; } # Call from pbx to SBC # Get sbc ip from dispatcher list ds_select_dst("1","4"); $var(new_uri) = "sip:" + $tU + "@" + $dd + ":5060"; $ru=($var(new_uri)); route(RELAY); } When I send a call to the kamailio its routed correctly to the first pbx. But when I kill the asterisk process on the first pbx, an invite is still send by Kamailio to the first pbx. Maybe I am assuming incorrectly that when the dispatcher does not react to the ds_ping it will be set to "inactive" and not be selected by the ds_select_dst command... Or am I thinking to easy and do I need to configure some actions in the failure route? Pretty sure I'm doing something wrong, but cant see where and have not been able to find the information. Tips are appreciated. Rgds, Gertjan Wolzak

2 1

onsend_route showing wrong answer after forking
by Aymeric Moizard 26 Sep '18

26 Sep '18

Hi, I'm experimenting "onsend_route" with "onsend_route_reply=yes" to improve my logs. I was surprised to see that "onsend_route" doesn't always display the "forwarded" answer after negative final answers. The documentation seems to indicate that this route will execute only if "the received reply is sent out". -as I'm forking, the final 6xx answer is not "forwarded" immediatly... and I guess this is the reason it doesn't show. Here comes a real strange use-case: With forking enabled, only the last answer will appear in the "onsend_route", however, in some case, this is NOT the answer being forwarded. Example: 1/ receive 603 decline from branch1 2/ a CANCEL is sent on branch2 3/ a 487 Request Terminated is received on branch2 What happens: 1/ onsend_route is showing "487 Request Terminated" 2/ 603 Decline is forwarded (as expected) The documenation say: "If set to 1 (yes, on), onsend_route block is executed for received replies that are sent out. Default is 0." This doesn't look to be correct. (either the doc, or the code?) Is there any possibility to install a log in the script that would log ALL *exact* replies being sent: 1/ the real forwarded ones 2/ and also the ones sent by kamailio directly. May be this is possible with some "event_route"? Tks a lot! Aymeric -- Antisip - http://www.antisip.com

1 0

kamailio 5.0.6 | segfault bad msg or missing FROM header
by Laurent Schweizer 26 Sep '18

26 Sep '18

Dear all, I have setup few weeks ago a kamailio 5.0.6, was working correctly till last night where 2 time he stop with a segfault error : Sep 25 23:36:19 ch0012 kamailio[18696]: ERROR: <core> [core/parser/parse_from.c:53]: parse_from_header(): bad msg or missing FROM header Sep 25 23:36:19 ch0012 kernel: kamailio[18696]: segfault at 0 ip 00000000004d7c4f sp 00007fff16135130 error 4 in kamailio[400000+42b000] Sep 25 23:36:19 ch0012 kernel: kamailio[18682]: segfault at 0 ip 00000000004d7c4f sp 00007fff16135130 error 4 in kamailio[400000+42b000] Sep 25 23:36:19 ch0012 kamailio[18696]: ERROR: siptrace [siptrace.c:535]: sip_trace_prepare(): cannot parse FROM header Sep 25 23:36:19 ch0012 kamailio[18682]: ERROR: <core> [core/parser/parse_from.c:53]: parse_from_header(): bad msg or missing FROM header Sep 25 23:36:19 ch0012 kamailio[18682]: ERROR: siptrace [siptrace.c:535]: sip_trace_prepare(): cannot parse FROM header Sep 25 23:36:19 ch0012 abrt-hook-ccpp: Process 18696 (kamailio) of user 0 killed by SIGSEGV - ignoring (repeated crash) Sep 25 23:36:19 ch0012 abrt-hook-ccpp: Process 18682 (kamailio) of user 0 killed by SIGSEGV - dumping core Sep 25 23:36:20 ch0012 abrt-hook-ccpp: Saved core dump of pid 18696 to core.18696 at / (2294079488 bytes) Sep 25 23:36:20 ch0012 kamailio[18759]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 31 Sep 25 23:36:20 ch0012 kamailio[18679]: ALERT: <core> [main.c:743]: handle_sigs(): child process 18696 exited by a signal 11 Sep 25 23:36:20 ch0012 kamailio[18679]: ALERT: <core> [main.c:746]: handle_sigs(): core was generated Sep 25 23:36:20 ch0012 kamailio[18679]: INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD Sep 25 23:36:20 ch0012 kamailio[18759]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18754]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18758]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18757]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18753]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18752]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18756]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18751]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18746]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18747]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18750]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received I have exported from the core file some info (file is attached), any idea ? BR Laurent

2 2

Dispatcher Questions
by Duarte Rocha 26 Sep '18

26 Sep '18

Greetings, I have some questions about Dispatcher0s modules general behaviours. Can Dispatcher use two tables to read values from? If I can't use two tables, is it possible to read them from a Mysql View? I want to be able to have a Gateway belonging to more than one Gateway Set. By doing so, in the table, the gateway will be repeated but with different groups. How will the probing work in this scenario? Will the Gateway be probed multiple times? Thanks for your help. Cheers

2 1

TM Module timing question
by Igor Olhovskiy 25 Sep '18

25 Sep '18

Hi! A bit lost in timers for TM module Idea is following I want to have dial timeout for 30 seconds, but only if I received 100 Trying in reply to INVITE. If no 100 Trying is received - I want timeout to be like 2 seconds. 100 Trying is indicator for me, that other side is alive. As I got, fr_timer and fr_inv_timer is not what I’m looking for. Regards, Igor

3 3

htable event routes behavior
by Володимир Іванець 25 Sep '18

25 Sep '18

Hello all, I have following configuration: 1. Htable which loads from MySQL DB and has autoexpire setting. *modparam("htable", "htable", "tl_connect_users=>size=4;autoexpire=120;dbtable=some_htable")* 2. Regular *route[SOME_ROUTE]* with sole logic. 3. *event_route[htable:mod-init]* 4. *event_route[htable:expired: some_htable]* I use while loop in *event_route[htable:mod-init]* to extract stored values from htable, assign them to *$var(some_variable)* and call *SOME_ROUTE* to perform some action for all of them on Kamailio start up. *SOME_ROUTE* has *$sht(some_htable =>$var(some_variable)) = "enabled"* line to add expired value back to the htable. The pboblem is when* event_route[htable:expired: some_htable]* triggers *SOME_ROUTE*. Values are not being added to htable. Hovewer if I will try to manipulate values for key_name or key_value by adding some character new item will be insert to the htable. Does anybody know if behavior is correct when exactly same value can not be inserted back to htable on it's expired event? Thanks a lot!

1 0

Help with kamailio/freeswitch
by David Villasmil 25 Sep '18

25 Sep '18

Hello guys, I know this is kamailio's mailing list, but I'm getting no answer on freeswitch, and i thought someone here has probably done this is the past. I have a proxy in front of freeswitch. I want freeswitch to use the proxy for everything including the termination to my provider. If I configured the gateway like this: <include> <gateway name="myprovider"> <param name="username" value="username" /> <param name="auth-username" value="username"/> <param name="password" value="password" /> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="from-domain" value="myprovider.com"/> <param name="realm" value="myprovider.com"/> <param name="register-proxy" value="myproxy.domain.com"/> </gateway> </include> The registration works properly, the REGISTER goes to the proxy like REGISTER sip:myprovider.com;transport=udp SIP/2.0 and the proxy forwards it fine, and registration works great. But when i try to call via this gateway, the call goes STRAIGHT to the provider! If, on the other hand, i configure my gateway as: <include> <gateway name="myprovider"> <param name="username" value="username" /> <param name="auth-username" value="username"/> <param name="password" value="password" /> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="from-domain" value="myprovider.com"/> <param name="realm" value="myprovider.com"/> <param name="register-proxy" value="myproxy.domain.com"/> <param name="proxy" value="myproxy.domain.com"/> </gateway> </include> Registration does NOT work, because the uri is: REGISTER sip:myproxy.domain.com;transport=udp SIP/2.0 Because there is no registrar on that proxy!

3 2

escape character
by David Villasmil 25 Sep '18

25 Sep '18

Hello guys, I'd like to search for " ' or & But i can't seem to figure out how... I tried: $tU =~ "[\"'&]" But that's not catching it... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Substitution Regex
by Duarte Rocha 24 Sep '18

24 Sep '18

Greetings, I'm using subst_hf() function to do some search and replace and i want to identify a undifined amount of digits after a string (ex: string=123123). Everywhere i look, the right way seems to be \d so i'm doing subst_hf("Via","/;string=\d/substitution_string/", "a"); It doesn't seem to work. Any ideas why? Best Regards, Duarte Rocha

3 2

Issuse with http_client and config_file
by Barry Flanagan 24 Sep '18

24 Sep '18

Hi I am trying to use http_client with a config_file set but can't get it to work .Have tried on 4.4 and 5.1 with the same results (Debian 9 packages). Here are my modparams modparam("http_client", "config_file", "/etc/kamailio/httpconnections.cfg") ..and here is the contents of httpconnections.cfg [apiserver] url = https://my.api.com/v1 username = apiuser password = api_password timeout = 1 authmethod = 2 maxdatasize = 4 tlsversion = TLSv1.2 verify_peer = yes Whenever I try to run this, I get the following error: http_client [curlcon.c:142]: curl_get_connection(): curl_get_connection no success in looking for httpcon: [apiserver] I am running the folowing: $var(res) = http_connect("apiserver", "/wlrmap/$var(wlrUserFromPAI).json", "$avp(wlrdata)"); >From the debug logs it appears that Kamailio is initialising the module OK (included below) so I am not sure why it is failing to find it subsequently. Any help appreciated! Thanks -Barry Flanagan Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: <core> [core/sr_module.c:988]: init_mod(): http_client Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:254]: mod_init(): init curl module Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:799]: curl_init_con(): curl_init_con httpcon: [apiserver] ID 3613147223 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:841]: curl_init_con(): CURL: Added connection [apiserver] Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:722]: fixup_raw_http_client_conn_list(): cname: [apiserver] url: [vapi-de v.op.telinsta.net/operator/v1] username [api%40operator-dev.telinsta.net] password [k3645QA36VtvsSLgo5m9Y827NCeKfPjhRnd879zx] failover [] timeout [1] useragent [kamailio (5.1.5 (x86_64/linux))] maxdatasize [0] Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:728]: fixup_raw_http_client_conn_list(): cname: [apiserver] client_cert [ (null)] client_key [(null)] ciphersuites [(null)] tlsversion [6] verify_peer [0] verify_host [0] keep_connections [0] Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:297]: mod_init(): **** init http_client module done. Curl version: 7. 52.1 SSL OpenSSL/1.0.2l Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:299]: mod_init(): **** init http_client: Number of connection objects : 1 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:301]: mod_init(): **** init http_client: User Agent: kamailio (5.1.5 (x86_64/linux)) Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:303]: mod_init(): **** init http_client: HTTPredirect: 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:305]: mod_init(): **** init http_client: Client Cert: Key Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:308]: mod_init(): **** init http_client: CA Cert: (null) Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:309]: mod_init(): **** init http_client: Cipher Suites: Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:311]: mod_init(): **** init http_client: SSL Version: 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:312]: mod_init(): **** init http_client: verifypeer: 1 verifyhost: 2 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:314]: mod_init(): **** init http_client: HTTP Proxy: Port 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:317]: mod_init(): **** init http_client: Auth method: 2 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:318]: mod_init(): **** init http_client: Keep Connections open: 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:321]: mod_init(): **** Extra: Curl supports TLS IPv6 IDN Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: <core> [core/route.c:872]: fix_actions(): fixing http_connect()

1 0

Module RTPPROXY param Timeout_Socket not Working
by Hamid Hashmi 24 Sep '18

24 Sep '18

Kamailio didn't start listening on IP mentioned in Module RTPPROXY parameter Timeout_Socket<https://kamailio.org/docs/modules/5.1.x/modules/rtpproxy.html#rtpproxy.p.ti…>. Below is my configuration file. loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_tout", 1) modparam("rtpproxy", "rtpproxy_retr", 2) modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2") loadmodule "xmlrpc.so" modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "mode", 1) modparam("xmlrpc", "url_match", "^/RPC2") when I type a command $ netstat -nlp it only shows SIP listening ports. Kamailio should also start listening on 127.0.0.1:8000. Below are the RTPproxy logs, RTPproxy is also unable to bridge the media DBUG:get_command:GLOBAL: received command "19083_4 Uc120,101 9a1060af660d4202976c419ab5300470 58.65.176.42 4000 1ea06d1eb0a8443697a08e568045598c;1" INFO:rtpp_command_ul_handle:GLOBAL: new session 9a1060af660d4202976c419ab5300470, tag 1ea06d1eb0a8443697a08e568045598c;1 requested, type strong INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new session on a port 35012 created, tag 1ea06d1eb0a8443697a08e568045598c;1 INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: pre-filling caller's address with 58.65.176.42:4000 DBUG:rtpc_doreply:GLOBAL: sending reply "35012 51.38.201.3 " DBUG:get_command:GLOBAL: received command "19083_5 Uc120,101 9a1060af660d4202976c419ab5300470 58.65.176.42 4022 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 1ea06d1eb0a8443697a08e568045598c;1 xmlrpc:http://127.0.0.1:8000/RPC2" INFO:rtpp_command_ul_handle:GLOBAL: new session 9a1060af660d4202976c419ab5300470, tag 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 requested, type strong INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new session on a port 35014 created, tag 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 ERR:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: invalid socket name 1ea06d1eb0a8443697a08e568045598c;1 INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: pre-filling caller's address with 58.65.176.42:4022 DBUG:rtpc_doreply:GLOBAL: sending reply "35014 51.38.201.3 " DBUG:get_command:GLOBAL: received command "19086_4 D 9a1060af660d4202976c419ab5300470 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq 1ea06d1eb0a8443697a08e568045598c" INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully deleting session 1 on ports 35012/0 INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 35012/0 is cleaned up INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully deleting session 1 on ports 35014/0 INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 35014/0 is cleaned up DBUG:rtpc_doreply:GLOBAL: sending reply "0 " DBUG:get_command:GLOBAL: received command "19086_5 D 9a1060af660d4202976c419ab5300470 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq 1ea06d1eb0a8443697a08e568045598c" INFO:handle_command:GLOBAL: delete request failed: session 9a1060af660d4202976c419ab5300470, tags 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq/1ea06d1eb0a8443697a08e568045598c not found DBUG:rtpc_doreply:GLOBAL: sending reply "E50 But if I comment out the param "Timeout_Socket" then there is no issue in bridging the media. Regards Hamid R. Hashmi

3 6

uac usage
by David Villasmil 21 Sep '18

21 Sep '18

Hello guys On my working config, i enabled uac like so: loadmodule "uac.so" ... modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_contact_addr", "test-proxy.singlecomm.com") ... And i added a user to the uacreg table: id,l_uuid,l_username,l_domain,r_username,r_domain,realm,auth_username,auth_password,auth_ha1,auth_proxy,expires,flags,reg_delay 1,10001,myuser,test-proxy.test.com,myuser,myprovider.com,myprovider.com ,myuser,mypass,,sip:myprovider.com,300,0,0 But it's not trying to register, and when i do: #> kamcmd uac.reg_info l_uuid 10001 I get: error: 404 - Record not found What am I doing wrong? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Jsonrpc_exec does not get body of request
by Denys Pozniak 21 Sep '18

21 Sep '18

Hello! I need to get some data via RPC to Kamailio inside by external event. But it seems long values of body brakes something: loadmodule "xhttp.so" loadmodule "jsonrpcs.so" modparam("jsonrpcs", "transport", 0) 1) event_route[xhttp:request] { ... jsonrpc_exec('{"jsonrpc": "2.0", "method": "core.psx", "id": 1}'); xlog("L_ERR","RPC code: $jsonrpl(code) \n"); xlog("L_ERR","RPC text: $jsonrpl(text) \n"); xlog("L_ERR","RPC body: $jsonrpl(body) \n"); return; } But in result I do not have body: Sep 21 14:27:55 /usr/sbin/kamailio[16140]: ERROR: <script>: RPC code: 200 Sep 21 14:27:55 /usr/sbin/kamailio[16140]: ERROR: <script>: RPC text: OK 2) If I request core.uptime I have something in body: event_route[xhttp:request] { ... jsonrpc_exec('{"jsonrpc": "2.0", "method": "core.uptime", "id": 1}'); xlog("L_ERR","RPC code: $jsonrpl(code) \n"); xlog("L_ERR","RPC text: $jsonrpl(text) \n"); xlog("L_ERR","RPC body: $jsonrpl(body) \n"); return; } Sep 21 14:37:17 /usr/sbin/kamailio[22683]: ERROR: <script>: RPC code: 200 Sep 21 14:37:17 /usr/sbin/kamailio[22683]: ERROR: <script>: RPC text: OK Sep 21 14:37:17 /usr/sbin/kamailio[22683]: ERROR: <script>: RPC body: {"jsonrpc":"2.0","result":{"now":"Fri Sep 21 14:37:17 2018","up_since":"Fri Sep 21 14:37:16 2018","uptime":1},"id":1} [root@kamailio]# kamailio -v version: kamailio 5.1.2 (x86_64/linux) 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_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. -- BR, Denys Pozniak

1 0

Include parameters in replies
by Duarte Rocha 21 Sep '18

21 Sep '18

Greetings, I need to to include some routing information (Two Id's, example "id_in=10;id_out=72") to be read later by Kamailio in every Request (Except the first one and in every Reply. For the requests, i'm including this info as parameters in the Record-Route and i'm reading it later in the Route Headers. For replies i thought about using the Via header. I'm using subst_hf() and i'm having success adding the parameter to Via in the requests. However, it does not seem the best idea since i breake the routing when i add the parameter. I wanted to avoid using custom headers for this since i can't really be sure that they won't be erased by any client's CPE or endpoint. Any ideas on this ? Thanks Cheers, Duarte Rocha

1 0

usr location
by volga629＠networklab.ca 21 Sep '18

21 Sep '18

Hello Everyone, User location lookup looks like can't handle long $rU like 10102-ce72256df4945bc472ed9c27a1037f46. It always return -3 404 not found. here relevant log Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: [INVITE] Call going from LAN SourceIP <10.18.130.50> to WAN. Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Looking user --> 10102-ce72256df4945bc472ed9c27a1037f46 in db Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Got [INVITE] and status <null> route[LOCATION] { $avp(oexten) = $rU; xlog("L_INFO", "Looking user --> $rU in db\n"); if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: route(MSG_STORE); # Meaning SMS has being stored for offline delivery if(!t_check_status("202")) { xlog("L_INFO", "Got [$rm] and status [$rs]\n"); # For rest traffic proceed as normal send_reply("404", "Not Found"); } exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } } kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 volga629

4 3

Re: [SR-Users] Daemon won't start
by Henning Westerholt 20 Sep '18

20 Sep '18

Am Samstag, 15. September 2018, 20:00:39 CEST schrieb hexhaxtron: > I can't make it work... > > It seems to have created the databases without any problem. > [pastebin cfg and logs] Hi Hexatron, please keep the list in CC. you should investigate why this error happens: Sep 14 06:52:07 thunix.org mysqld[1943]: 2018-09-14 6:52:07 140187532334848 [Warning] Access denied for user 'kamailio'@'localhost' (using password: YES) Sep 14 06:52:07 thunix.org /usr/sbin/kamailio[4254]: ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) Can you login with a mysql client on a shell to the database with the same credentials? Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

1 0

Kamailio On Tap – Social Event In Dusseldorf, Sep 27, 2018
by Daniel-Constantin Mierla 20 Sep '18

20 Sep '18

Hello, we are organizing an open social networking event in the evening of Sep 27, 2018, in Dusseldorf, at the end of the first day of Kamailio Developers meeting. Anyone in the area that wants to enjoy beer, pizza and chat about Kamailio, VoIP/RTC is welcome to join us. More details at: * https://www.kamailio.org/w/2018/09/kamailio-on-tap-social-event-in-dusseldo… Note that participation is free, but you have to announce your participation until Sep 26 in order to plan properly the event. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 0

usr location
by volga629＠networklab.ca 20 Sep '18

20 Sep '18

Hello Everyone, User location lookup looks like can't handle long $rU like 10102-ce72256df4945bc472ed9c27a1037f46. It always return -3 404 not found. here relevant log Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: [INVITE] Call going from LAN SourceIP <10.18.130.50> to WAN. Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Looking user --> 10102-ce72256df4945bc472ed9c27a1037f46 in db Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Got [INVITE] and status <null> route[LOCATION] { $avp(oexten) = $rU; xlog("L_INFO", "Looking user --> $rU in db\n"); if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: route(MSG_STORE); # Meaning SMS has being stored for offline delivery if(!t_check_status("202")) { xlog("L_INFO", "Got [$rm] and status [$rs]\n"); # For rest traffic proceed as normal send_reply("404", "Not Found"); } exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } } kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 volga629

1 0

re- double record route
by Wilkins, Steve 19 Sep '18

19 Sep '18

Hi Henning, Yes I do have that enabled. What is happening is that one of the providers on IOS is sending a double record route on the INVITE, but it is getting lost somewhere so when I send a 'BYE', I get a "404 not here". When I look at the SIP message I see only one of the record routes from the INVITE, therefore my assumption is, this is what is causing the 404, because everything else looks good.

3 8

occasional processing delay and UAC remote registration
by Dmitri Savolainen 19 Sep '18

19 Sep '18

Hi all. i encountered in my 4.1.5 kamailio server some inconstant issue. I have 2 listen address: addr1 for incoming SIP requests and as source addr for remote registration (via UAC module) addr2 for incoming SIP requests only uac config for regs: modparam("uac", "reg_keep_callid", 1) modparam("uac", "reg_random_delay", 120) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_timer_interval", 40) i have response time checking tool, that send OPTIONS every second, which immedeately replied by sl_send_reply("200") without any processing. Usualy respose time <1ms. Sometimes (2-3 per day) addr1 stop to response for these OPTIONS (timeout >5sec), while addr2 continue responding with time <1ms. In kamailio.log at this moment bunch of messages like ERROR: uac [uac_reg.c:1034]: uac_reg_tm_callback(): got sip response 403 > while registering ERROR: uac [uac_reg.c:1034]: uac_reg_tm_callback(): got sip response 408 > while registering That is normal: some remotes are broken. The problem is: 1. usually these messages appearing is not the reason of problem, but all problems are while these messages in log 2. it may be on OFF peak / ON peak day period 3. in my test installation i have much more remote failure registration (X 10), and the problem is not manifested more often (but i can catch it 2-3 per day) I think adding more workers for addr1 may help (now 8 worker and 16 CPUs). But it is interesting to understand the source of this issue -- Savolainen Dmitri

1 2

Where does location ruid come from?
by David Cunningham 19 Sep '18

19 Sep '18

Hello, We are debugging an issue with location records having duplicate "ruid" values, which aren't allowed by the table index. Can anyone tell us where the "ruid" value comes from? Is it a function of some other variable? Thanks in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

Can't get from cache: $sqlrows(asg)
by David Villasmil 18 Sep '18

18 Sep '18

Hello guys, i've got this in my config: Module config: modparam("sqlops","sqlcon","asg=>mysql://DB_USER:DB_PASS@DB_HOST/kamailio") Route: sql_query("asg", "insert ignore dispatcher (id, setid, destination, flags, priority, attrs, description) values (NULL, $var(dispatcher_id), 'sip:$si:5080', 0, 0, 'weight=$var(new_weight);maxload=50', '$fU')"); xlog("L_ERR", "[ASG_ADD_TO_DISPATCHER] Affected rows: $sqlrows(asg)\n"); if ($sqlrows(asg)>0) { ds_reload(); } else { xlog("L_ERR", "[ASG_ADD_TO_DISPATCHER] COULD NOT ADD THE NEW GATEWAY WITH IP $si AND USERNAME $fU\n"); } and i'm getting: 0(3929) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1261, column 20-32: Can't get from cache: $sqlrows(asg) I can't figure out where the problem might be. Can anyone help? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 4

Await to INVITE after REGISTER for mobile device which get awake via android/iOS notification messages
by Kristijan Vrban 18 Sep '18

18 Sep '18

Hi, now that many SIP endpoints are mobile devices which usually deep-sleep, and only get awake after Android/iOS own notification messages awake them, what are the best practices to await a a pending INVITE to such a mobile device until it REGISTER after the awake? I currently use a Asterisk a do a simple loop, trying new INVITES every second, but this is not elegant. A nice solution would be a type of queue, where the pending INVITE get send to the new location triggered immediately by the fresh REGISTER from that device to avoid any delay during call setup Kristijan

2 1

Incoming call on Kamailio WebRTC
by Adesh Pandey 17 Sep '18

17 Sep '18

Hi All, We are trying to setup WebRTC with Kamailio as sip registrar and dispatcher. We have succeed for outgoing calls but cannot attend incoming calls to the Kamailio. It would be a great help if someone can provide a sample config for webrtc + kamailio or blog for the same. Thanking you all -- Adesh Pandey Sr. Software Developer [image: phone] +91 92129 92129 Ext: 56 [image: moible] +91 8527384897 [image: Profile Pic] [image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter] <https://twitter.com/MyOperator>[image: Twitter] <https://www.youtube.com/user/MyOperatorCo>[image: Linkedin] <https://www.linkedin.com/company/voicetree-technologies>[image: Google+] <https://plus.google.com/+MyoperatorCo/posts> [image: Logo] <http://myoperator.co/>

2 1

Re: [SR-Users] Record-Route Parameters
by Duarte Rocha 17 Sep '18

17 Sep '18

Hey, I need all of my kamailio's IPs in Record Routes and i can do that already. What i want to do is add different parameters to each Record-Route. Is that possible? A sexta, 14/09/2018, 17:54, Duarte Rocha <duarterocha91(a)gmail.com> escreveu: > Greetings, > > My Kamailio has 2 IP's and, as such, it will generate two Record-Routes > when the Request is relayed. > > I want to add one parameter to one of the Record Routes and another > parameter to the other Record-Route. However, if i use add_rr_param() it > will automatically add the parameters to the two Record-Route Headers. > > Is there a way to do what i want without me re-writing the headers > directly ? > > Cheers, > > Duarte Rocha >

3 8

how to catch attacker using bad request line?
by Juha Heinanen 17 Sep '18

17 Sep '18

In order to be able to fail2ban an attacker that sends tons of SIP requests with malformed request lines, it would need to be possible to generate an appropriate syslog message from config file. I didn't find any sanity module param value that would turn on checking of request line syntax. Any other ideas? As an example, below is what comes to syslog when I send a request that has syntax error on request line. -- Juha Sep 17 14:46:39 char /usr/bin/sip-proxy[9458]: ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [(null)] Sep 17 14:46:39 char /usr/bin/sip-proxy[9458]: WARNING: <core> [core/receive.c:230]: receive_msg(): parsing relevant headers failed Sep 17 14:46:43 char /usr/bin/sip-proxy[9458]: ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [(null)] Sep 17 14:46:43 char /usr/bin/sip-proxy[9458]: WARNING: <core> [core/receive.c:230]: receive_msg(): parsing relevant headers failed

2 11

issue with rewrite to domain in failure route
by Yuriy Gorlichenko 17 Sep '18

17 Sep '18

Hi I'm building some system with failover trunk but my providers requires to set their doamin in domain part of To Header so when I rewrite it at the my failure_route $td = FAILOVER_PROVIDER my To filed looks like To: <sip: 123(a)my.failover.providermy.main.provider> what i tried: Move this change to BRANCH_ROUTE: here tried as $td = as uac_replace_to('',"newToFieldValue) Move to the my custom route with both variants But result is the same kamailio 5.1 version: kamailio 5.1.5 (x86_64/linux) d06080 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d06080 compiled on 00:43:08 Aug 23 2018 with gcc 6.3.0

5 11

Dispatcher Active IP for specific SetID over xHTTP
by Soltanici Ilie 17 Sep '18

17 Sep '18

How could I get the active IP from the dispatcher module for a specific setid by using xHTTP request?

1 0

double record route from INVITE
by Wilkins, Steve 16 Sep '18

16 Sep '18

Hi Mojtaba, But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE' (from IOS) need to be there, so that IOS can find it's Proxies and complete the transaction and send back a '200 OK'? Thank you,

1 0

double record route?
by Wilkins, Steve 16 Sep '18

16 Sep '18

Good Morning All, Is there any way to add a double record route? I tried adding a second record route and I always only get the first one added. I have tried record_route(....), and record_route_advertised_address(...), but I still only get the first record route added. Thank you, -Steve

2 1

Allow registration only for one client per AOR
by eyas barhouk 15 Sep '18

15 Sep '18

Hello dears, I'm using kamailio devel version as an IMS (P-CSCF,S-CSCC,I-CSCF) with forking mode ,when more than one client registered with the same AOR and the servers received a call to that Aor All the clients ringing , after that one of the clients answer the call which will cause cancel message with reason call completed elsewhere . This cancel message affect on the running call which answered by one of the client and terminate it . So how to set up kamailio to allow registration only for one client per Aor or to prevent the affection of this cancel messages . Thanks in advance & best regards

4 5

kamailio installable on stretch?
by David Villasmil 15 Sep '18

15 Sep '18

Hello guys, is kamailio from deb http://deb.kamailio.org/kamailio51 stretch main deb-src http://deb.kamailio.org/kamailio51 stretch main Installable on debain stretch? I'm getting: # apt-get install kamailio kamailio-xmlrpc-modules kamailio-mysql-modules Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'kamailio-xml-modules' instead of 'kamailio-xmlrpc-modules' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio : Depends: libreadline7 (>= 6.0) but it is not installable kamailio-mysql-modules : Depends: libmariadbclient18 (>= 5.5.36) but it is not installable E: Unable to correct problems, you have held broken packages. # but libreadline7 is not avaiable on stretch. Thanks David Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 3

Daemon won't start
by hexhaxtron 15 Sep '18

15 Sep '18

Hi! I did this tutorial: https://sonnguyen.ws/install-kamailio-debianubuntu/ When I try to start the daemon I get: # systemctl start kamailio Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. Doing systemctl status kamailio shows: # systemctl status kamailio ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Fri 2018-09-14 20:26:25 BST; 44s ago Process: 29482 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: 24331 (code=exited, status=0/SUCCESS) Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: usrloc [dlist.c:654]: register_udomain(): failed to open database connection Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: registrar [reg_mod.c:566]: domain_fixup(): failed to register domain Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: <core> [route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:646 Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: <core> [route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:649 Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Sep 14 20:26:25 thunix.org systemd[1]: kamailio.service: Control process exited, code=exited status=255 Sep 14 20:26:25 thunix.org systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Sep 14 20:26:25 thunix.org systemd[1]: kamailio.service: Unit entered failed state. Sep 14 20:26:25 thunix.org systemd[1]: kamailio.service: Failed with result 'exit-code'. My /etc/kamailio/kamailio.cfg is: http://paste.debian.net/1042245/ This is the output of journalctl --no-pager|grep kamailio http://paste.debian.net/1042246/ What should I do to fix this? Thanks, hexhaxtron

2 1

Record-Route Parameters
by Duarte Rocha 15 Sep '18

15 Sep '18

Greetings, My Kamailio has 2 IP's and, as such, it will generate two Record-Routes when the Request is relayed. I want to add one parameter to one of the Record Routes and another parameter to the other Record-Route. However, if i use add_rr_param() it will automatically add the parameters to the two Record-Route Headers. Is there a way to do what i want without me re-writing the headers directly ? Cheers, Duarte Rocha

2 1

Record Route Parameters
by Duarte Rocha 15 Sep '18

15 Sep '18

Greetings, I have Kamailio as an SBC with mutiple IP's and I need to save some information in the first Invite about the routing made internally in the Kamailio. I'm thinking about adding some parameters to Record-Route and retrieve them in the Route header. Is this safe ? Can I be sure that those parameters won't be deleted or changed? Best Regards, Duarte Rocha

3 2

maxbuffer value
by Dmitri Savolainen 14 Sep '18

14 Sep '18

Hi. Does anybody know what kamailio core maxbuffer parameter means? And is there any conjunction with system net.core.rmem_max parameter? Doc is obscure for me "The size in bytes not to be exceeded during the auto-probing procedure of descovering the maximum buffer size for receiving UDP messages. " -- Savolainen Dmitri

2 1

kamailio installable on stretch?
by David Villasmil Govea 14 Sep '18

14 Sep '18

Hello guys, is kamailio from deb http://deb.kamailio.org/kamailio51 stretch main deb-src http://deb.kamailio.org/kamailio51 stretch main Installable on debain stretch? I'm getting: # apt-get install kamailio kamailio-xmlrpc-modules kamailio-mysql-modules Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'kamailio-xml-modules' instead of 'kamailio-xmlrpc-modules' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio : Depends: libreadline7 (>= 6.0) but it is not installable kamailio-mysql-modules : Depends: libmariadbclient18 (>= 5.5.36) but it is not installable E: Unable to correct problems, you have held broken packages. # but libreadline7 is not avaiable on stretch. Thanks David -- DVG -- Imagination is more important than knowledge Albert Einstein ᐧ

1 0

Forward SIP request between two Kamailio servers
by Nathanael Eneroth 14 Sep '18

14 Sep '18

Hi,? I have a simulation setup of 4 VMs in Virtual Box - two UAs and two Kamailio servers. The first UA will register to KAM1, and the second UA will regsiter to KAM2. See the Figure below. UA1<--->KAM1<---->KAM2<---->UA2 int1 int1 int3 int3 int2 int2 I would like KAM1 to forward all SIP requests destined to int3 via KAM2 and vice versa. I am aware of the routing logic in 'kamailio.cfg', but i find rather excessive. Pseudo code of what i'm trying to achieve: if (is_method("INVITE")) { if(dst_ip == int3) set_next_kamailio_server(KAM2) } Or: if (is_method("INVITE")) { if(callee is unkown) ask_kamailio_server(KAM2) } Thanks for all the help

4 3

DMQ - Failed to get domain
by Daniel W. Graham 14 Sep '18

14 Sep '18

Receiving the following message on the node receiving usr_loc via DMQ, for each location received: DEBUG: dmq [worker.c:87]: worker_loop(): dmq_worker [0 54958] lock acquired DEBUG: dmq_usrloc [usrloc_sync.c:460]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@x.x.x.x DEBUG: dmq_usrloc [usrloc_sync.c:405]: usrloc_dmq_execute_action(): Received DMQ_UPDATE. Update contact info... ERROR: dmq_usrloc [usrloc_sync.c:70]: add_contact(): Failed to get domain Any input is appreciated. Thanks, -dan

2 10

Restart kamailio on critical error
by Sergey Safarov 14 Sep '18

14 Sep '18

Is possible to restart kamailio when is critical errors is happens? Now in log ``` Sep 13 18:50:38 ca-edm-sw3.phonebox.work sh[1171]: 81(85) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/85 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:743]: handle_sigs(): child process 5 exited by a signal 9 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:743]: handle_sigs(): child process 80 exited by a signal 9 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:743]: handle_sigs(): child process 110 exited by a signal 9 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:746]: handle_sigs(): core was not generated ``` Some of childs is died but main kamailio process is still active. SIP traffic is not processed because TCP socket is died. When i core is generated i can force restart process. But i cannot parse logs to catch such event. May exist other way? Sergey

2 2

Python KEMI performance
by Aleksandar Sosic 14 Sep '18

14 Sep '18

Hi all, is there anyone here that has done a performance testing measurements between native kamailio config file and full python kemi instance of kamailio? I've read somewhere that the python kemi code was 30% slower. Why is that and are there any plans on trying to improve the performance of python kemi? Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

3 2

limit call per duration
by António Silva 13 Sep '18

13 Sep '18

Hi, I'm new to kamailio, i want to be able to limit call per duration, how can i do it? i found module call_control, http://www.kamailio.org/docs/modules/5.1.x/modules/call_control.html, do you recommend it to use? anyone using it that would like to share the experience, like the values for cpu usage, is it realtime, support for multiple calls per account, other... Thanks!! -- Saludos / Regards / Cumprimentos António Silva

1 2

How to use Geoip Module in Kemi
by Naveed Ur Rehman 13 Sep '18

13 Sep '18

Hi, I am trying to use geoip2 module in kemi using LUA, The issue is that when i try to access the $gip2(src=>cc) variable using lua, the kamailio instance killed and stops, this is my code snippet, kindly advise anyone how this can be sorted local sourceIP=KSR.pv.get("$si"); if KSR.geoip2.match("sourceIP","src") then local x= KSR.pv.get("$gip2(src=>cc)"); end thanks Engineer Naveed Ur Rehman 0923415652825 Computer Systems Engineering

2 1

TOPOH module im REGISTER method.
by Mojtaba 13 Sep '18

13 Sep '18

Hello, I am used topoh module in kamailio, Why this module does not work (No change ) in Register request Method? I need to change UA' ip address in Via header in Register request before forwarded to my network? Does the topoh module used in this situation? I have not this issue in INVITE request, I just have this issue in Register request. Thanks. With Regards.Mojtaba -- --Mojtaba Esfandiari.S

3 5

Crashed drouting
by Евгений Голей 12 Sep '18

12 Sep '18

Good evening. I have errors on the server (kamailio v 5.1.0) "kamailio [30529]: segfault at 0 ip 00007f9c00c29cfb sp 00007fffa9808310 error 4 in drouting.so [7f9c00c26000 + 39000]" which cause the service to stop. Although the drouting module has been used for a long time, this has never happened. This was done after adding the functions of the dispatcher module. What could be the reason ? Example error in system kamailio [20728]: segfault at 0 ip 00007f32c1c8fcfb sp 00007fff78430d00 error 4 in drouting.so [7f32c1c8c000 + 39000] in the logs of kamailio 018-09-08T22: 30: 29.0448 "ALERT: <core> [main.c: 746]: handle_sigs (): child process 10556 exited by a signal 11" 2018-09-08T22: 30: 29.0452 "ALERT: <core> [main.c: 749]: handle_sigs (): core was generated" 2018-09-08T22: 30: 29.0456 "INFO: <core> [main.c: 771]: handle_sigs (): terminating due to SIGCHLD" -- B.R. Evgeniy

3 7

remove_hf("Bla") for all requests
by Sergiu Pojoga 12 Sep '18

12 Sep '18

Hi there, Say I need to remove_hf("User-Agent") for all requests, back and forth. So I add it at the top of *request_route *section. However, replies don't seem to be affected by it. Do I really need to arm a *t_on_reply route* for this simple purpose? "Problem" with that is that *route[RELAY]* already has some *onreply_route* block doing things like *NATMANAGE *for some methods and as far as I know - only one* onreply_route* can be armed for a request? route[RELAY] { ... if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } t_relay(); } onreply_route[MANAGE_REPLY] { ... route(NATMANAGE); } Any suggestions? Thanks.

4 7

Re: [SR-Users] kamailio + rtpengine + custom filename
by Жан Базаров 12 Sep '18

12 Sep '18

Message-ID: <CABVi_EzZRF8nUqoBGcUtUeAV2pknn5S0urcmJFgHSSz6N1kJ5g(a)mail.gmail.com> Content-Type: text/plain; charset="UTF-8" Hello, You could use metadata to save your favourite filename on it. After recording is finished, replace it. The metadata file is created based on <CALL_ID> for each call. With Regards. Mojtaba On Mon, Sep 10, 2018 at 8:53 PM Жан Базаров <chiefkeeft(a)gmail.com> wrote: > > Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> > How i can change filename in start_recording() ??? Thx you. Can you help me with it task? Show me, please, an example. I want filename of the following format, for example: 'output-dir/yyyy/mm/dd/fromnumber/callid-tonumber-uniqueid.wav'.

2 1

Missing ACC record in case of canceled call forking
by Marco Capetta 12 Sep '18

12 Sep '18

Hi All, I'm facing a strange problem of missing ACC record in case of parallel call forking. The scenario is the following: - A subscriber with 1 device registered - B subscriber with 2 device registered (B1 and B2) CASE 1: - A calls B - B1 and B2 start ringing - A hangups the call before B1 or B2 answers Kamailio generates an ACC record. CASE 2: - A calls B - B1 and B2 start ringing - B1 rejects the call sending back a 486 - B2 is still ringing - A hangups the call before B2 answers Kamailio DOESN'T generate an ACC record. We have Kamailio v5.1.4 with TM module enabled. ACC configuration is the following: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) I increased debug level of TM and ACC modules and I added some debug lines as well. The difference between the two calls is that, after the CANCEL sent by A is processed by Kamailio, in CASE 1 I have the following lines: Aug 28 14:21:11 spce proxy[13188]: DEBUG: tm [t_hooks.c:258]: run_trans_callbacks_internal(): DBG: trans=0x7f5a88049198, callback type 512, id 0 entered Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_logic.c:670]: tmcb_func(): acc callback called for t(0x7f5a88049198) event type 512, reply code 487 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 0 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 1 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 2 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 3 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 4 Those debug lines are not printed for CASE 2. Is there any configuration I'm missing or is it a bug? Thank you very much for your support. Marco

2 14

Call termination via dialog RPC dlg.end_dlg for the SIP-UA behind NAT
by Denys Pozniak 11 Sep '18

11 Sep '18

Hello! When calling RPC command dlg.end_dlg, BYE for leg behind NAT sends according to the Contact header not to the real ip:port. RPC ul.dump shows correct address in Received: sip:2.2.2.2:7488, but looks dialog does not know real address. How can I force sending to the real address? [root@proxy1]# kamcmd dlg.list { h_entry: 3753 h_id: 6741 call-id: 0_618427980(a)10.10.10.10 from_uri: sip:1234567890@1.1.1.1:5060 to_uri: sip:3333333333@1.1.1.1:5060 state: 2 start_ts: 0 init_ts: 1536671492 timeout: 0 lifetime: 7200 dflags: 512 sflags: 4096 iflags: 32 caller: { tag: 2750096738 contact: sip:1234567890@10.10.10.10:5060 cseq: 2 route_set: socket: udp:1.1.1.1:5060 } callee: { tag: contact: cseq: route_set: socket: } profiles: { } variables: { } } [root@proxy1]# kamcmd ul.dump Info: { AoR: 1234567890 HashID: 262653578 Contacts: { Contact: { Address: sip:1234567890@10.10.10.10:5060 Expires: 1269 Q: -1.000000 Call-ID: 0_213504837(a)10.10.10.10 CSeq: 8 User-Agent: Yealink SIP-T21P Received: sip: 2.2.2.2:7488 Path: [not set] State: CS_NEW Flags: 0 CFlags: 65600 Socket: udp: 1.1.1.1:5060 Methods: 16383 Ruid: uloc-5b97c358-6052-1 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1536674159 Last-Modified: 1536674159 } } } -- BR, Denys Pozniak

1 0

Remove branch from branch list
by Yuriy Gorlichenko 11 Sep '18

11 Sep '18

Hi I'm trying to manipulate with branches at the $branch variable. 8 actually need to remove unneeded branch if not passed some checks. For now I Just rewriting uri and dst_uri with some face address but it looks little bit dirty way. Is here any an other more clear way to do this?

2 2

Record Routes
by Wilkins, Steve 11 Sep '18

11 Sep '18

Hello All, Can a Route: be removed?, if so, how? Thank you?

2 4

Logging of messages generated by Kamailio itself
by Konstantin Polyakov 11 Sep '18

11 Sep '18

Hello, Could somebody help with one issue? In some cases Kamailio can send messages itself. For example, when fr_inv_timer is expired Kamailio sends 408 to the INVITE sender and CANCEL to the INVITE receiver. These two messages (408 and CANCEL) are not printed into the log and I cannot find a way to do that. There are also other scenarios when some Kamailio modules send SIP messages by its own. So is there way to print out into log all such messages? My goal is to get full message flow in the log per certain call-id. Thank you in advance. Best regards. Konstantin

2 9

kamailio + rtpengine + custom filename
by Жан Базаров 11 Sep '18

11 Sep '18

Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> How i can change filename in start_recording() ???

2 1

tpops module
by volga629＠networklab.ca 11 Sep '18

11 Sep '18

Hello Everyone, I am using tcpops module for keepalive and on BYE when tcp keeplaive close is called produce this error in log. Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: <core> [core/sr_module.c:1781]: get_int_fparam(): Could not convert PV to int Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: tcpops [tcpops_mod.c:293]: w_tcp_keepalive_disable1(): invalid parameter 'con' (must be a number) kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 if (is_method("BYE|CANCEL")) { $avp(bye_conid) = $conid; rtpengine_delete(); t_on_reply("4"); } onreply_route[4] { if(is_method("BYE") && t_check_status("200")) { tcp_keepalive_disable("$avp(bye_conid)"); } } volga629

3 3

Simultaneous usage of t_on_reply() and t_on_failure()
by Володимир Іванець 10 Sep '18

10 Sep '18

Hello everyone, I wanted to ask those who know if Kamailio's behavior I'm facing is expected or I should make some improvements to the configuration. Kamailio version is 5.1.0. I have a route where RTPEngine parameters are being collected and *rtpengine_offer()* is called. After that *t_on_reply("REPLY_SIP_TO_SIP");* followed by the *t_on_failure("FAILURE_SIP_TO_SIP");* are used. The idea is to process all responces except 415 or 488 from UAC as usual in *onreply_route[REPLY_SIP_TO_SIP]* and use *failure_route[FAILURE_SIP_TO_SIP]* to update SDP with *rtpengine_offer()* if necessary. *onreply_route[REPLY_SIP_TO_SIP]* just goes to *exit;* if *$rs* equals 415 or 488. This works fine with Htek phone which sends 100, 180 and then 488. But I can not see *failure_route[FAILURE_SIP_TO_SIP]* execution for calls to Zoiper which replies with 100 and immediately 415. t_on_failure(failure_route) documentation says: "Sets failure routing block, to which control is passed after a transaction completed with a negative result but before sending a final reply." and to be honest I don't really understand how lacking of responce prevents failure_route from executing. Thanks a lot!

4 5

Crashed kamailio drouting
by Евгений Голей 08 Sep '18

08 Sep '18

Good evening. I have errors on the server (kamailio v 5.1.0) "kamailio [30529]: segfault at 0 ip 00007f9c00c29cfb sp 00007fffa9808310 error 4 in drouting.so [7f9c00c26000 + 39000]" which cause the service to stop. Although the drouting module has been used for a long time, this has never happened. This was done after adding the functions of the dispatcher module. What could be the reason ? Example error in system kamailio [20728]: segfault at 0 ip 00007f32c1c8fcfb sp 00007fff78430d00 error 4 in drouting.so [7f32c1c8c000 + 39000] in the logs of kamailio 018-09-08T22: 30: 29.0448 "ALERT: <core> [main.c: 746]: handle_sigs (): child process 10556 exited by a signal 11" 2018-09-08T22: 30: 29.0452 "ALERT: <core> [main.c: 749]: handle_sigs (): core was generated" 2018-09-08T22: 30: 29.0456 "INFO: <core> [main.c: 771]: handle_sigs (): terminating due to SIGCHLD" -- BR Evgeniy

1 0

kamailio.cfg errors with this example from documentation
by Wilkins, Steve 07 Sep '18

07 Sep '18

$var(x) = "abcd"; $(var(x){s.substr,1,0}); Hello all, I took the example from the documentation, but Kamailio.cfg has errors with this example. Errors=> 0(14444) CRITICAL: <core> [core/cfg.y:3489]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 1-23: pvar with transformations in assignment left side 0(14444) CRITICAL: <core> [core/cfg.y:3492]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: syntax error 0(14444) CRITICAL: <core> [core/cfg.y:3492]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: bad command Thank you,

2 5

Via header line missing
by Karlheinz Knapp 07 Sep '18

07 Sep '18

Hi, hm.... the only reason why I'm playing with kamailio is that it should be able to translate IPv6 from customer phones behind Fritz!Box (NAT) to IPv4 used by the provider (which is not able to handle IPv6). But it looks that he accept in general the packet with 2 Via lines (also with IPv6 as address) and send a normal reply. The second Via is not needed, because he have to send to the first Via. And if it really comes to trouble, I can remove it with kamailio :) If it is not possible to make a complete translation with kamailio, then I have to search again for an other solution. But still: why is this one Via line not transmitted on the IPv6 side? Only when I do some modifications it appears magically. That's not a normal behaviour in my opinion. If you or Henning can not reproduce this behaviour, then it's possible that I start TeamViewer and I can show this on our installation (only at normal working hours). Btw. I use sngrep to display all the messages, but it works stable and reliable. Best regards, i.A. Bernd

1 0

Event: Kamailio Developers Meeting, Sep 27-28, 2018, in Dusseldorf, Germany
by Daniel-Constantin Mierla 07 Sep '18

07 Sep '18

Kamailio SIP Server project is organizing a meeting of its developers during September 27-28, 2018, hosted by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join. Please note we have a limited capacity of 20 seats in the meeting room. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: - https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 1

kamailio with evapi crashing on high volume
by Jayesh Nambiar 07 Sep '18

07 Sep '18

Hello, I'm using kamailio 5.1.5 with evapi. I have a node.js connecting with kamailio evapi to which I send events and also consume events based on which I do the routing. I have 8 evapi workers defined in the config. The problem is that kamailio randomly crashes on high load. I'm assuming that it is related to the evapi module as rest of the config is pretty straight forward. I could get a core file and here's the core dump: [New LWP 14042] [Thread debugging using libthread_db enabled] Using host libthread_db library “/lib/x86_64-linux-gnu/libthread_db.so.1”. Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/siptrunk.pid -f /usr/local/carrie’. Program terminated with signal SIGABRT, Aborted. #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54 54 ../sysdeps/unix/sysv/linux/raise.c: No such file or directory. (gdb) bt #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54 #1 0x00007f999528502a in __GI_abort () at abort.c:89 #2 0x000000000041a029 in sig_alarm_abort (signo=14) at main.c:646 #3 <signal handler called> #4 0x00007f999534f497 in __libc_cleanup_routine (f=<optimized out>) at ../sysdeps/nptl/libc-lockP.h:291 #5 closelog () at ../misc/syslog.c:415 #6 0x0000000000000000 in ?? () Any help in this regards is would allow me to identify the reason of the crash. Thanks for the support. - Jayesh

2 5

Kamailio with CGNAT
by Amar Tinawi 06 Sep '18

06 Sep '18

Hello Although my question it's not related to Kamailio directly, but i think someone could help in this strange behavior in our tests, we have some clients under private networks, where the provider of those networks implements CGNAT (NAT444) for it's customers to reach internet (Client Private IP) --->(Home router: Private to Private NAT)--->(FW with CGNAT)---------------------------->(Kamailio Proxy Public IP) Registration is performing fine, but making calls from this client had strange problem as follow: Client -------------INVITE--------------> Proxy Proxy -------------Trying---------------->Client Proxy--------------INVITE---------> Called Party The called party should send Ringing msg, but the massage get lost somewhere and didn't reach to the Proxy. Although all the IP's in sip massages are public i tried with STUN/TURN and NAThelper module solutions, but didn't help is there any way to slove CGNAT with SIP in Kamailio? Thanks in Advance

4 11

dlg.end_dlg on RTP timeout
by Igor Olhovskiy 06 Sep '18

06 Sep '18

Hi all! Is there any way to end dialog in Kamailio on RTP Timeout event on RTPEngine? Or only look at logs/redis database with external tool? Regards, Igor

5 13

call_stats
by David Villasmil 06 Sep '18

06 Sep '18

Hello all, Anyone got the table definition for call_stats? I'm getting db_mysql_submit_query(): driver error on query: Table 'kamailio.call_stats' doesn't exist (1146) And i can't find the table anywhere. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 2

Via header line missing
by Karlheinz Knapp 06 Sep '18

06 Sep '18

Hi, of course kamailio is listening on IPv4 and IPv6. Register coming from IPv6 and goes out on IPv4: REGISTER sip:sip.ggew-net.de SIP/2.0 Via: SIP/2.0/UDP 217.113.176.155;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 Via: SIP/2.0/UDP [2A05:E2C2:0:401:F40D:E8C5:5D85:2A95]:5060;received=2A05:E2C2:0:401:F40D:E8C5:5D85:2A95;branch=z9hG4bK001755af58b0e811a048bb9f393b8bfe;r rt=5060 From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de> Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER Contact: <sip:4962519448281@ [2A05:E2C2:0:401:F40D:E8C5:5D85:2A95]:5060>;+sip.instance="<urn:uuid:00E0F96C-5BAF-E811-A877-86B14B4F5249>" Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK Max-Forwards: 69 Allow-Events: org.3gpp.nwinitdereg User-Agent: SIPPER for PhonerLite Supported: replaces, from-change, gruu Expires: 900 Content-Length: 0 P-hint: outbound Answer on IPv4: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 The same answer on th IPv6 side after kamailio without something special: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 It contains no Via header!!! If I add my append_hf() append_hf("Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=$sel(via[1].branch);rport=5060\r\n", "To"); msg_apply_changes(); I get: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 With the original Via header !!! If I add a second Via with append_hf() append_hf("Via: SIP/2.0/UDP [2a05:e2c2:0:401:3002:AA:754A:F475]:5060;rport\r\n", "To"); append_hf("Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=$sel(via[1].branch);rport=5060\r\n", "To"); msg_apply_changes(); I can see this additional line in the IPv6 packet too. SIP/2.0 401 Unauthorized Call-ID: 0038C332-5AB0-E811-ABC9-80F5AA6CB72C@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=2618219059 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07871-795b4386-0dc31d606 Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=z9hG4bK0ce8.b94e5d3389049004487a8551ac2c8c41.0;rport=5060 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK0ce8.b94e5d3389049004487a8551ac2c8c41.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="795b3345266c3dd57a52e3237f4276dd",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 But since I don't have the right branch, PhonerLite starts a new registration instead of reregister with username password. Best regards

2 1

null $au
by David Villasmil 06 Sep '18

06 Sep '18

Hello, in the block: if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } I'm trying to print out $au, but it's coming up NULL many times, i don't get why this is. If i remember correctly, the script goes through this block twice, once when we get the REGISTER with no credentials, and once when the REGISTER comes with credentials after the challenge... is this correct? i need to know how to catch $au when the authentication is successful... help? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 2

Via header line missing
by Karlheinz Knapp 05 Sep '18

05 Sep '18

Hi, we want to use kamailio as a IPv6 to IPv4 proxy. IPv4 to IPv4 bridging with NAT works. If we come from an IPv6 address, the registration packet is send via IPv4 to the provider. He answers with 401 Unauthorized and one Via line in the header. On the IPv6 side was no Via line in the packet. So I tried to add one line by hand using an onreply_route and append_hf(), msg_apply_changes() To my suprise, this line does never appear, but the original Via line is now in the packet !!! If I add now additional Via header lines, they appear in the packet too. This looks realy like a bug to me. A solution is needed, because the phone does not respond to the 401 with user and password. It starts a new registration instead. Maybe I can fix it with my additional Via linem but I need the original branch ID (I think) How can I get this? Btw. we have this behaviour with 4.4.2 and 5.1.5 If I get no answer during the next 12 hours I'll try 3.3.7 Best regards

2 1

dispatcher local socket
by Евгений Голей 05 Sep '18

05 Sep '18

Hello! Kamilio version 5.1.0 Please tell me why the module dispatcher does not correctly determine the availability of destination gateways. The scheme is as follows: Kamailio (udp:XX.XX.XX.XX:5060) ----- Provider_1 -----> (sip:aa.bb.cc.dd:5060) Dest GW Kamailio (udp:YY.YY.YY.YY:5060) ------- Provider_2 -----/ In table dispatcher: setid | destination | flags | priority | attrs 17 | sip:aa.bb.cc.dd:5060 | 8 | 50 | socket=udp:XX.XX.XX.XX:5060 17 | sip:aa.bb.cc.dd:5060 | 8 | 40 | socket=udp:YY.YY.YY.YY:5060 That is, our server controls the availability of the remote gateway through two local sockets, in order to switch to the backup if it is not available through the primary Internet provider. But as I understand at the moment when determining the availability of a remote gateway, the outgoing socket does not play any role, that is, if the remote gateway is not accessible through the secondary operator (priority is less than the primary one), only the one that has priority above is always inactive. Are there any solutions to this problem? BR. Evgeniy

1 0

From URI attribute for dispatcher target
by Giacomo Vacca 05 Sep '18

05 Sep '18

Hi all, I was looking for a solution to set a dedicated From URI in OPTIONS requests to a specific dispatcher target. ds_ping_from ( https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc…) applies globally, why I needed the granularity of a single target. With these changes to the dispatcher module I get what I need: https://github.com/kamailio/kamailio/commit/3f9c38be11416f7abe5c4448088ba66… I can just add an attribute to the others, e.g. instead of *1 sip:127.0.0.1:7070 <http://127.0.0.1:7070> 0 1* something like: *1 sip:127.0.0.1:7070 <http://127.0.0.1:7070> 0 1 ping_from=sip.custom.com <http://sip.custom.com>* When the ping_from attribute is not specified, the general ds_ping_from applies and is used. This would save me from a solution like intercepting the outgoing OPTIONS and manipulating the From header. Does this approach make sense? Are there other ways already available to achieve this? Thanks, Giacomo

1 0

SCA: sca_call_info_bye_handler: no in-use callee appearance for BYE
by Asgaroth 05 Sep '18

05 Sep '18

Hi All, I've just started playing with the SCA module and am encountering an issue whereby SCA cannot seem to find an in-use appearance for the BYE message. From my limited understanding of SCA, the module, in this case, appears to be trying to track the state of the call using callid as there are no call-info headers in the signalling for it to extract and use. I call sca_call_info_update() for initial invite, 100 response, 180 response, 200 response and again for the terminating bye. When I answer the call, at the 200 ok for the initial invite, I can see sca sending a notify to the subscribed devices indicating appearance state being active. I can verify this with kamcmd sca.all_appearances. However, when I hang up the call, when sca_call_info_update() is fired for the bye message it fails with the following error message: sca [sca_call_info.c:1587]: sca_call_info_bye_handler(): sca_call_info_bye_handler: no in-use callee appearance for BYE <aor> from <from uri>, call-ID 7989c223-2ba4-1237-1888-20474786734e I cannot seem to track down where this is failing for me, or where I may be missing something in my configuration. sca.all_appearances does appear to have the entry mentioned in the failure message, there must be something I'm missing somewhere. Is there anyone out there who is using the sca module and may have a moment to assist with this issue. I have a pcap trace that I can send privately. Any assistance would be greatly appreciated. Thanks

1 0

Kamailio v5.1.5 Released
by Daniel-Constantin Mierla 04 Sep '18

04 Sep '18

Hello, Kamailio SIP Server v5.1.5 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.5. For more details about version 5.1.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/08/kamailio-v5-1-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 1

Problem with reconnection to database server
by Minh Phan 04 Sep '18

04 Sep '18

Hi List, My Kamailio instance is disconnected from database server (Postgres) quite often (after running for some days) and it won't manage to reconnect. Any idea why this happens? Normally it try to reconnect when connection with the DB is lost, desn't it? Bellow are the related log messages. Best regards, Minh Phan Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR, Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:609]: db_postgres_insert(): unexpected result returned Sep 4 11:31:20 : ERROR: topos [tps_storage.c:599]: tps_db_insert_dialog(): failed to store message Sep 4 11:31:20 : ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK66fa.3f007792000000000000000000000000.0' Sep 4 11:31:20 : ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: insert into topos_d (rectime,a_callid,a_uuid,b_uuid,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag,a_rr,b_rr,s_rr,iflags,a_uri,b_uri,r_uri,a_srcaddr,b_srcaddr,s_method,s_cseq ) values ('2018-09-04 11:31:20','1ff8d1d7606f7416-26611(a)192.168.70.106 ','','','','','','','ae86e22a93159b5c6ac75234cdf26c1b-44e5','','','','',0,'','','','','','OPTIONS','10') Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 kamailio[26608]: ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK86fa.424ded00000000000000000000000000.0' Sep 4 11:31:20 kamailio[26608]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 kamailio[26608]: ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR, Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:609]: db_postgres_insert(): unexpected result returned Sep 4 11:31:20 : ERROR: topos [tps_storage.c:599]: tps_db_insert_dialog(): failed to store message Sep 4 11:31:20 : ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK76fa.d33cc801000000000000000000000000.0' Sep 4 11:31:20 : ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: insert into topos_d (rectime,a_callid,a_uuid,b_uuid,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag,a_rr,b_rr,s_rr,iflags,a_uri,b_uri,r_uri,a_srcaddr,b_srcaddr,s_method,s_cseq ) values ('2018-09-04 11:31:20','1ff8d1d7606f7417-26611(a)192.168.70.106 ','','','','','','','ae86e22a93159b5c6ac75234cdf26c1b-917f','','','','',0,'','','','','','OPTIONS','10') Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR,

2 1

DMQ mem leak issues
by Rogelio Perez 04 Sep '18

04 Sep '18

Hello, We're running three instances of Kamailo v5.14 as registrars handling registrations from ~2000 SIP clients, with one instance being primary and the other two as backups. The three of them are using the dmq and dmq_usrloc modules to synchronize user locations, however after a couple of days of operation the two failover instances show memory leak behaviors, with mem usage assigned to the core taking all available resources. When this happens we've noticed that: - The shared memory used by the function "sip_msg_shm_clone" spikes (from 1kb to 1.5GB). - The shared memory used by the function "dmq:worker.c:job_queue_push" also increases, but not as much (from 1kb to 1MB) - DMQ request are not being answered (with a 200 OK) by the affected instance during this memory leak, which make us think that DMQ module becomes unresponsive. A few more notes: - The failover instances are doing nothing except receiving replicated contacts. - The shared memory grows at the same rate on both instances, but the critical behavior never happens at the same time. - We are allocating 1GB memory on startup to each instance. - We store the location DB in a psql DB and we load it at startup. - We didn't find any errors in syslog, even at debug level. Has anyone experienced a similar issue who can suggest a possible solution? Thanks, Rogelio Perez Telnyx

6 23

Ipv6_to_ipv4
by Karlheinz Knapp 04 Sep '18

04 Sep '18

i have tested the sample config from: https://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 i found that i can register with v4 to kamailio and to the provider if i user ipv6 to kamailio and ipv4 to provider i can´t register. the packet from the provider contain a via entry which is missing from kamailio to my phone. Someone have a solution ?

1 0

event_route [tcp:closed] not executed
by Juha Heinanen 04 Sep '18

04 Sep '18

Based on comment: https://github.com/kamailio/kamailio/issues/171#issuecomment-418175840 I gave event_route [tcp:closed] a try: event_route [tcp:closed] { # Handle TCP connection close xlog("L_INFO", "$proto connection closed ($conid)\n"); } Nothing comes to syslog when I quit my baresip UA that according to wireshark closes its TCP connection after un-registration (fin - fin - ack). I have not set closed_event param, because according to README, it defaults to 1.

2 10

dispatcher probe
by volga629＠networklab.ca 04 Sep '18

04 Sep '18

Hello Everyone, Is possible access OPTIONS packet generated by dispatcher probe ? I am trying insert something like this. if(is_method("OPTIONS") && $hdr(User-Agent)=~"kamailio") { append_hf("X-Node-ID: $avp(node_id)\r\n"); } volga629

3 4

Can't connect to local MySQL server (13 "Permission denied")
by Ginhoux, Patrick 04 Sep '18

04 Sep '18

Hi, In a previous post ([SR-Users] Questions about changes between 5.0.1 and 5.0.7 <https://lists.kamailio.org/pipermail/sr-users/2018-August/102442.html> ), I reported problem to upgrade my kamailio from 5.0.1 to 5.0.7-7 level. With help from some people, it was finally fixed on my test server in my lab. Now I was expecting to do the same upgrade on a real server using these packages: kamailio-5.0.7-7.el7.x86_64.rpm kamailio-ims-5.0.7-7.el7.x86_64.rpm kamailio-mysql-5.0.7-7.el7.x86_64.rpm kamailio-perl-5.0.7-7.el7.x86_64.rpm kamailio-snmpstats-5.0.7-7.el7.x86_64.rpm But kamailio failed to connect the local MySQL server : Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: <core> [db.c:209]: db_bind_mod(): using db bind api for db_mysql Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: <core> [db.c:314]: db_do_init2(): connection 0x7fef779958f8 not found in pool Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: db_mysql [km_my_con.c:98]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/siprouter Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (13 "Permission denied") Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: htable [ht_db.c:92]: ht_db_open_con(): failed to connect to the database Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: <core> [core/sr_module.c:974]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) My "kamailio.cfg" contains the following command: modparam("htable", "db_url", "mysql://siprouter:password@localhost/siprouter") The service unit : "/usr/lib/systemd/system/kamailio.service" content is : [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=network.target mysql.service [Service] Type=simple User=kamailio Group=kamailio Environment='CFGFILE=/etc/kamailio/kamailio.cfg' Environment='SHM_MEMORY=1024' Environment='PKG_MEMORY=8' EnvironmentFile=-/etc/sysconfig/kamailio ExecStart=/usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY Restart=on-failure [Install] WantedBy=multi-user.target The "/etc/sysconfig/kamailio" content is : # # Kamailio startup options # # # Kamailio startup options # # Set to yes to enable kamailio, once configured properly. RUN_KAMAILIO=yes # User to run as USER=kamailio # Group to run as GROUP=kamailio # Amount of shared memory to allocate for the running Kamailio server (in Mb) SHM_MEMORY=1024 # Amount of per-process (package) memory to allocate for Kamailio (in Mb) PKG_MEMORY=8 Below the existing user in mysql : mysql:root:(none)> use mysql; Database changed mysql:root:mysql> select user,host from user; +-----------+-----------+ | user | host | +-----------+-----------+ | myadmin | % | | siprouter | % | | kamailio | localhost | | root | localhost | +-----------+-----------+ I have a symlink to locate the my.cnf : ln -s /myqdata/mysiprdb/data/my.cnf /etc/my.cnf And also this link to fix a htable reload problem : ln -s /myqdata/mysiprdb/data/mysiprdb.sock /var/lib/mysql/mysql.sock I don't find what the problem is this time. So if there is people that could help on the issue, thanks in advance. Cordialement. Patrick GINHOUX

3 6

load-balance to registered trunks?
by David Villasmil 03 Sep '18

03 Sep '18

Hello guys, I've got this scenario where some gateways will register on my kamailio and i need to send them calls on a load-balance fashion. is this possible? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

4 6

Dispatcher question
by Igor Olhovskiy 03 Sep '18

03 Sep '18

Hi! I’m having some issues with dispatcher and cannot get why it’s not working Idea is to use ds_next_dst right after ds_select_dst I’m having it like this modparam("dispatcher", "list_file", "/tmp/dispatcher.list") modparam("dispatcher", "flags", 2) # Enable dispather failover modparam("dispatcher", "xavp_dst", "_dispacher_dst_list_") … ds_select_dst(«1», «0»); … # Some extra check on $du here … ds_next_dst(); But get in logs WARNING: dispatcher [dispatch.c:2285]: ds_update_dst(): no xavp uri field in next destination record But contents of /tmp/dispatcher.list is 1 sip:172.28.0.12:5060 1 sip:172.28.0.19:5060 Can I somehow dump xavp_dst var to check if ds_select_dst is correct? kamailio 5.2.0-dev6 Regards, Igor

5 17

handle_ruri_alias() question or issue?
by Wilkins, Steve 02 Sep '18

02 Sep '18

Good Morning All, The following Incoming request came in => SIP Incoming Request: [[BYE sip:2406506175@20.20.20.20:5060;alias=10.10.10.5~55157~2 SIP/2.0 I called handle_ruri_alias(), and expected the destination and port to be set to 10.10.10.5:55157, but it was not. It was left 20.20.20.20:5060. Is my thinking correct or is there another way to set the destination and port to the alias? Thanks All!, -Steve

3 9

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). Any help would be massively appreciated. The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK<sip:proxy@%7bASTERISK> EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}<sip:%7bCLIENT%20IP%7d>>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}<sip:proxy@%7bPROXY%20IP%7d>> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY<sip:%7bPROXY> IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}<sip:proxy@%7bASTERISK%20INTERNAL%20IP%7d>>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}<sip:%7bPROXY%20IP%7d>> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060<sip:asterisk@%7bASTERISK%20EXTERNAL%20IP%7d:5060>> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

CentOS_7/kamailio_v5.1.x-rpms/ [Errno -1] Package does not match intended download.
by Denys Pozniak 31 Aug '18

31 Aug '18

Hello! Please check repository for Kamailio v5.1.x for CentOS7: http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… I got error: kamailio-debuginfo-5.1.5-1.el7 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not SatisfiableB --:--:-- ETA Trying other mirror. kamailio-outbound-5.1.5-1.el7. FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. kamailio-5.1.5-1.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-debuginfo-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-outbound-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. -- BR, Denys Pozniak

2 1

Re: [SR-Users] Final reply code is wrong in case of 302 and 486 busy
by Marco Capetta 31 Aug '18

31 Aug '18

Hi Daniel, did you have some time to have a look at the issue? Thank you very much Regards Marco

2 1

How to detect and Hangup a call on network lost at Caller or Callee?
by Hamid Hashmi 31 Aug '18

31 Aug '18

Dear Users Please advise how to detect a network lost at caller or callee side. I have tried timeout_socket<https://kamailio.org/docs/modules/5.1.x/modules/rtpproxy.html#rtpproxy.p.ti…> and keepalives <https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dialog.p.ka_int…> method but nothing worked for me. RTPproxy did not send timeout notification to Kamailio (Issue #1597<https://github.com/kamailio/kamailio/issues/1597>) while there is no options packet within a dialog. Is there any other method to cater such scenario. Regards Hamid R. Hashmi

5 11

Quick way to enable/disable all uac outbound registrations
by Joel Serrano 31 Aug '18

31 Aug '18

Hi, On an active/passive kamailio cluster, we use keepalived to move a virtual IP between the nodes and all signaling goes to that VIP. One of our DID providers requires us to register with every single DID we have from them in order to receive calls for such DID (NOTE: we only want to have the "active" kamailio sending outbound registrations to the carrier). My idea was to have keepalived enable all the outbound registrations when the current node changes to MASTER and disable them when it changes to BACKUP. Is there a quick "enable all" or "disable all" for uac module outbound registrations or do I have to "know in advanced" all of them and enable/disable one by one? In the docs we have: *8.4. uac.reg_disable* *Disable a remote registration record based on a filter. The command has two parameter: attribute and value. The attribute can be: l_uuid, l_username, r_username or auth_username. The value is what should be matcheg against the value of the attribute in the remote registration record.* I don't know if there is an "all" or maybe if a wildcard would work, but from the description it doesn't seem to. Any ideas? Do you guys think this is taking a correct approach? All suggestions are more than welcome!! Thanks, Joel.

3 7

Request Line alases
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, If the below Requests from 40.40.40.40(Asterisk) is received by 10.10.10.10(Kamailio) Request Line: INFO sip:3145553313@10.10.10.10:5060;alias=30.30.30.30~43508~2 SIP/2.0 Shouldn't Kamailio forward this request to 30.30.30.30? so that the '200 OK' can sent back to Asterisk for the response. (In this scenario) Would it be ok to just force a '200 OK' back to Asterisk from Kamailio? Thank you,

2 1

Delay
by Daniel W. Graham 30 Aug '18

30 Aug '18

Recently encountered a delay issue, packets arrive on system but Kamailio doesn't process right away. Here are a few details, any direction on what I can do to narrow down is appreciated. The receive queue normally sits at 0 or has random spike, when this issue occurs receive queue remains high like the below output. Version 5.1.5 ss -4 -n -l | grep 5060 udp UNCONN 321536 0 207.241.130.2:5060 *:* tcp LISTEN 0 128 207.241.130.2:5060 *:* -dan

2 2

IPv6 2 Ipv4
by Karlheinz Knapp 30 Aug '18

30 Aug '18

Hi i serach for a config to convert IPv6 to ipv4. My Provider give me only IPv4 and i have 4000 Customer which have ipv6. Anybody have expierience and a running config ?

3 2

Interesting issue with ACK back from 200 OK INVITE
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, I have had an issue for quite some time where I needed to open two UDP ports (via LISTEN) for some softphone UAC's to stay connected. Funny enough there is one Provider that works with the Single UDP. I think I finally see why I was needing two listening UDP ports. When the ACK is returning from Asterisk to Kamailio, the ACK is coming back as protocol TCP, not SIP, causing the call to drop. I finally realized this but I am not sure what would cause this and why one provider phone works but not the others. Once this is resolved, I may actually say...It works! Thank you, -Steve

1 0

Notify: stateless error reply used: I'm terribly sorry, server error occurred
by Soltanici Ilie 29 Aug '18

29 Aug '18

For some reason, Voicemail Notify messages coming from Asterisk Boxes - are not forwarded to the SIP Extensions by Kamailio. This is the SIP Packet coming to the Kamailio: 2018/08/28 10:59:19.165644 1.1.1.1:5060 -> 2.2.2.2:5060 NOTIFY sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK5f628579;rport Max-Forwards: 70 Route: <sip:2.2.2.2:5060;transport=tcp;ftag=0472e823;lr=on;nat=yes> From: "asterisk" <sip:asterisk@1.1.1.1>;tag=as1833c329 To: <sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2>;tag=0472e823 Contact: <sip:asterisk@1.1.1.1:5060> Call-ID: 93183ZjE3YmI0OGYxMDNiMzhlNzVjZjRjNTc2NTMwZGNmYzk CSeq: 103 NOTIFY User-Agent: NUACOM Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@1.1.1.1 Voice-Message: 0/0 (0/0) And this is the message which I'm receiving in the Kamailio logs: [RELAY] Error when forwarding Request ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) This is the Relay route: # Forwarding Statefully if ( ! t_relay()) { xlogl( "L_ERR" , "[RELAY] Error when forwarding Request \n " ); sl_reply_error(); } exit ;

2 2

Info: Cluecon Weekly about Kamailio, today at 17:00GMT
by Daniel-Constantin Mierla 29 Aug '18

29 Aug '18

Hello, if you haven't seen the news article on kamailio.org, today at 17:00GMT I will be part of the Cluecon Weekly online video conference to present about what's new lately in Kamailio, plans for the future as well as address questions about using Kamailio together with FreeSwitch. Anyone can join via audio or video calls, more details about at: * https://www.kamailio.org/w/2018/08/kamailio-on-cluecon-weekly-aug-29-2018/ * https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 1

Kamailio consuming UDP packets very slowly leading to high RECV - Q
by sagar malam 29 Aug '18

29 Aug '18

Hello, I am using Kamailio as a SIP proxy.So it receives SIP packet from internet and forwards it to FS servers in local network.When i execute "ss" command i see very high value in RECV-Q column.I THINK IT IS NOT NORMAL.PLEASE CORRECT ME IF I AM WRONG. ==================================================== [root@fep-1 proc]# ss -u -a -n -e | grep 5060 UNCONN 0 0 10.50.8.1:5060 *:* ino:6831630 sk:fd <-> UNCONN 0 0 10.50.7.18:5060 *:* ino:6831629 sk:fe <-> UNCONN *1183104* 0 10.50.7.254:5060 *:* ino:6831627 sk:ff <-> UNCONN *84864* 0 2607:f900:1:3::254:5060 :::* ino:6831628 sk:100 v6only:1 <-> ====================================================== Initially i thought that there is something in script which must be causing kamailio to process UDP request slower but i faced same issue with a very simple script where i simply reply with stateless 200 OK for each sip request : request_route { sl_send_reply("200","OK");exit; $avp(uuid) = $rm + "-" + $ci; ........ ......... ......... ....... Server configuration : OS : CENTOS 7 Kernel : 4.16 CPU : 5 X Intel(R) Xeon(R) CPU E5-2680 v2 @ 2.80GHz RAM : 32 GB Please help me debugging this issue.Thanks in advance

4 9

Using SEMS in IMS-ZTE Network
by Mojtaba 28 Aug '18

28 Aug '18

Hello, I am working with IMS (ZTE vendor), I want to use SBC in front of IMS network for some security policy (Hide Topology), I used SEMS as SBC in this regards, (like what we have in IMS/SEMS in kamailio by Mr Carsten Bock). When the Sebscriber want to register, The Register request received by sems and the forward to IMS(ZTE), But the "503 Service Unavailable" is received from IMS. I enable "enable_reg_caching=yes" in register.sbcprofile''s configuration file in sems. If i use kamailio, instead of SEMS, The Registration request is done successfully.In Kamailio i just forward the Register request to IMS.like this: route { if (is_method("REGISTER")) { $du = "sip:172.23.1.1:5080"; route(RELAY); } } Let me know do the issue is from myself or from IMS(ZTE)? In the fillowing, I paste all SIP Register signalling. First in kamailio, second with SEMS. ================Kamailio================== 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5e58 -> 0x4193!] UDP, length 787 E../....@.]...............^XREGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 08:20:53.114198 IP (tos 0x0, ttl 255, id 25839, offset 0, flags [none], proto UDP (17), length 685) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 657 E...d......$..............l.SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipTR9m3KdZ*3-5-16648*daei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:20:53.120142 IP (tos 0x10, ttl 64, id 48874, offset 0, flags [none], proto UDP (17), length 1137) 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5f9a -> 0xcc62!] UDP, length 1109 E..q....@.\V.............]_.REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> To: <sip:+985137054518@r-kh.com;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",uri="sip:r-kh.com;transport=UDP",response="9469852a06c01efeabf3a7a5dd1a9f3b",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Content-Length: 0 08:20:53.203871 IP (tos 0x0, ttl 255, id 25842, offset 0, flags [none], proto UDP (17), length 1118) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 1090 E..^d......p.............J.5SIP/2.0 200 OK Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipjRydzgMa*3-5-16648*daej.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Contact: <sip:+985137054518@192.168.122.1:64407;transport=UDP;rinstance=e664e178fc12691e>;expires=1800 Contact: <sip:ZTEAGCF000f425704be720000@agcf2.r-kh.com:5079;appid=2-7;UserRegInfo=002-02-00-0b80-04-172.30.48.143;BACaddr=172.30.48.143;BACport=2944>;expires=1800;+sip.instance="<urn:uuid:172.30.35.41-000>";reg-id=1 P-Associated-URI: <tel:+985137054518> P-Associated-URI: <sip:+985137054518@r-kh.com> Date: Sat, 25 Aug 2018 12:25:17 GMT Authentication-Info: nextnonce="00cfbd64b136c12a4e27b67fd28d19fb",qop=auth,rspauth="91ca956c5eb6ee9f1934e2e770750077",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001 Content-Length: 0 ============================SEMS======================== 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5e16 -> 0xe11e!] UDP, length 721 E...C.@.@.................^. REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bK~0uk0ahN;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.790414 IP (tos 0x0, ttl 255, id 35790, offset 0, flags [none], proto UDP (17), length 593) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 565 E..Q.....................=..SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bK~0uk0ahN To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipDEmH2AWT*3-5-16648*egei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 CSeq: 10 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:23:32.855067 IP (tos 0x0, ttl 64, id 17364, offset 0, flags [DF], proto UDP (17), length 1071) 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5f58 -> 0xf553!] UDP, length 1043 E../C.@.@................._X REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bKeAfKkaZa;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",uri="sip:r-kh.com;transport=UDP",response="017efb986e17506a82bb01e8eb136b90",cnonce="3764fd5eb2ea9539d61ec14196a2bf00",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.860417 IP (tos 0x0, ttl 255, id 35804, offset 0, flags [none], proto UDP (17), length 508) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 480 E...........................SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bKeAfKkaZa To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipxxsD3FR2VG8e9zxR_1dx*1-1-16648*ihad.1 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 CSeq: 10 REGISTER X-ZTE-Cause: "CSCF-1.1.BC004016-BC004288-0-0.pcscf1.r-kh.com" Retry-After: 30 Content-Length: 0 -- --Mojtaba Esfandiari.S

2 3

Parmas in lcr_gw
by Ashutosh chaubey 28 Aug '18

28 Aug '18

Hi, How to give username and password for the gateway in params column in lcr_gw? Regards, Ashutosh

2 1

regular expression in kamailio
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi All, I am trying get the 10digit number called in on using the following $var(caller) = $(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]}); I thought I could use a regular expression for the expression in re.subst,expression Thank you, -Steve

2 8

Listner problems
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi Joel, Actually, I have a strange issue. If I have a single UDP IP:Port listening, most Providers/Phones have two-way Audio/Video, however, with some Providers/Phones, I need to have two UDP IP:Port listners in order to get two-way Audio/Video. Here is the strange thing. If I enable two, the Providers/Phones that worked when I had a single listener only have one-way Audi/Video. I use Kamailio 5.1 and Asterisk 15.3 (pjsip). This behavior is so strange, Thank you, -Steve

2 2

add listen dynamically
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi All, Is it possible to add or remove "listen" dynamically? Thank you, -Steve

2 2

Struggling with RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi Florian, Thank you for your response. I checked and direct-media is off. Just as a recap, here is where I am I just can't get rtpengine to work. I have tried multiple configurations, but to no avail. Note that calls work good if rtpengine is disabled. Here is my setup => Public IP: 20.20.20.20 Private IP 10.10.10.10 flow => webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip phone rtpenngine startup (I have tried a few different startups) => rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10\!20.20.20.20 --listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 10000 -M 20000 -E kamailio => Invites: rtpengine_manage("trust-address replace-origin replace-session-connection direction=ext direction=int ICE=remove RTP/AVP"); Reply's: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); I have tried direction ext ext; and many other combinations, each producing its own incorrect behavior. Thanks again, Steve

1 0

Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hello all, I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this? I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience. Thanks all, -Steve

8 17

Send a BYE for the call
by Nicolas Breuer 23 Aug '18

23 Aug '18

Hello, Any ideas how to cut a call in progress ? How to send a forced "BYE" to all parties ? Thanks

2 2

Kamailio 5.1.4 crashes
by Abdulaziz Alghosh 23 Aug '18

23 Aug '18

Hi Everyone, I am trying to start kamailio 5.1.4 with two own modules. After /etc/init.d/kamailio start , I am getting one error of the dispatcher module and bunch of warrnings because of missed reply and failure routes. The first error before crashing is: ERROR: dispatcher [dispatcher.c:814]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Later on, when I am trying to register a user (from a soft phone) i am getting the following: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 20 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:746]: handle_sigs(): child process 6942 exited by a signal 11 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:749]: handle_sigs(): core was not generated Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [main.c:771]: handle_sigs(): terminating due to SIGCHLD Aug 22 10:02:52 sgw1a /sbin/kamailio[6959]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6958]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6957]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6956]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6955]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6954]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6953]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6950]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6941]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6944]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6948]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6951]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6946]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6952]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6943]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6949]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6947]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6938]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6937]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6940]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6939]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a kernel: [296301.271457] kamailio[6942]: segfault at 0 ip (null) sp 00007fffb3ddf368 error 14 in kamailio[400000+442000] Aug 22 10:02:52 sgw1a /sbin/kamailio[6945]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6936]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_fifo.c:594]: jsonrpc_fifo_destroy(): cannot delete the fifo (/tmp/kamailio_jsonrpc.fifo): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_sock.c:511]: jsonrpc_dgram_destroy(): cannot delete the socket (/tmp/kamailio_rpc.sock): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regarding to the first error of the dispatcher module i could not find out which calibration i can follow with its parameters to get rid of this error. Regarding the jasonrpcs and the last two erros, here are my configuration in kamailio.cfg: #/* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "fifo_mode", 0755) modparam("jsonrpcs", "fifo_group", "kamailio") modparam("jsonrpcs", "fifo_user", "kamailio") #/* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") modparam("jsonrpcs", "dgram_mode", 0755) modparam("jsonrpcs", "dgram_user", "kamailio") modparam("jsonrpcs", "dgram_group", "kamailio") I think the root cause is even beyond the dispatcher and jsonrpcs modules setting. Please check the attached where you can find my complete log file after starting Kamailio. Best regards Abdulaziz

2 4

Dispatcher - Force one end point active
by Marrold 23 Aug '18

23 Aug '18

Hi, We are currently using the dispatcher module and probing end points with OPTIONS, and making them active / inactive depending on the response: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;code=404") modparam("dispatcher", "ds_ping_from", "[REDACTED]") modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_probing_mode", 1) One of our suppliers endpoints doesn't respond to SIP OPTIONS, is it possible to force a single endpoint as active, but continue probing the rest? I've tried various settings - both globally and flags for the individual end point, but I couldn't seem to find the correct combination. Thanks Matthew

2 2

Re: [SR-Users] Fail to reload htable
by Henning Westerholt 23 Aug '18

23 Aug '18

Am Montag, 20. August 2018, 20:58:43 CEST schrieb Soltanici Ilie: > In the logs, I see these errors: > > ERROR: <core> [db.c:463]: db_use_table(): invalid parameter value > Aug 20 19:56:41 kamailio[12716]: ERROR: <core> [db.c:463]: db_use_table(): > invalid parameter value Aug 20 19:56:41 kamailio[12716]: ERROR: htable > [ht_db.c:211]: ht_db_load_table(): failed to use_table Hi, please keep the list in CC. This error is related to a database issue. Please double check if there are any issues during startup in the log. Do you see any other error there? A similar (not 100% identical) issue was discussed recently here: https://lists.kamailio.org/pipermail/sr-users/2018-August/102393.html Maybe this helps you as well. Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

4 7

Re: [SR-Users] Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 23 Aug '18

23 Aug '18

Good morning Daniel, If RTPEngine is on the same server as Kamailio (Asterisk being on another server), and RTP traffic is sent to and from RTPEngine, then the provider only needs to whitelist one IP-Address. I thought with RTPEngine that all RTP traffic would go through it and then it would pass it on to the correct destination. Is this correct? Thank you -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Daniel Tryba Sent: Thursday, August 23, 2018 4:36 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine MITRE WARNING: Do not open unexpected password-protected attachments. >>>Email originates from a non-MITRE system. Use caution.<<< On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote: > The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something? In what sense do they need whitelisting? In a common NATed solution where is no white/blacklist needed. UA gets RTP endpoints from SDP, starts sending packets to ip/port and the destination will send back packets to the source ip/port, the router/firewall will just send this to the actual UA. I have yet to find an UA that cares about where the RTP stream is coming from with regards to the SIP traffic.

2 1

SIP and RTP Proxy for multiple Servers
by sewlist＠gmail.com 23 Aug '18

23 Aug '18

Im very new to kamailio, we have some ptime issues which causes one way audio due to varies PBX vendors connecting to our class 5 softswitches I would like to use Kamailio as the SIP/RTP proxy in front of our class 5 soft switches I followed the example of freeswitch and kamailio. However the config is still hard for me below setup works, however RTP not being proxied, Looks like its its not, because im not using NAT, so RTP goes direct https://freeswitch.org/confluence/display/FREESWITCH/Kamailio+basic+setup+a… I would like to to ensure RTP proxy is always used(possible re-framing ptime) and we have multiple soft switches Is it possible to setup kamailio.cfg so my customers can point the endpoint to kamailio and domain names contains the server we want to use to register with? eg, softswitch 1 = 1.1.1.2 softswitch 2 = 1.1.1.3 softswitch 3 = 1.1.1.4 etc etc my current config #!KAMAILIO #!define IPADDRESS "1.1.1.1" #!define SWITCH_IPADDRESS "1.1.1.2" #!define FLAG_FROM_SWITCH 1 #!define FLAG_FROM_USER 2 # ------------------ module loading ---------------------------------- loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "path.so" loadmodule "rtpproxy.so" # ----------------- setting module-specific parameters --------------- modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") # ------------------------- request routing logic ------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif # main routing logic route { # per request initial checks route(SANITY_CHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } route(CHECK_SOURCE_IP); ################################## ### HANDLE SEQUENTIAL REQUESTS ### route(WITHINDLG); ############################### ### HANDLE INITIAL REQUESTS ### t_check_trans(); if (is_method("INVITE|REFER")) { record_route(); } if (is_method("REGISTER")) { add_path(); } if (isflagset(FLAG_FROM_SWITCH)) { # don't send INVITE from SWITCH back to SWITCH, set reply route to handle NAT and forward them along t_on_reply("EXTERNAL_REPLY"); } else { # set destination to your SWITCH $du = "sip:1.1.1.2:5060"; } route(RELAY); } route[SANITY_CHECK] { if (!sanity_check()) { #xlog("L_WARN", "$ci|end|message is insane"); exit; } if (!mf_process_maxfwd_header("10")) { #xlog("L_WARN", "$ci|end|too much hops, not enough barley"); send_reply("483", "Too Many Hops"); exit; } if ($ua == "friendly-scanner" || $ua == "sundayddr" || $ua =~ "sipcli" ) { #xlog("L_WARN", "$ci|end|dropping message with user-agent $ua"); exit; } if ($si == IPADDRESS) { #xlog("L_WARN", "$ci|end|dropping message"); exit; } } route[CHECK_SOURCE_IP] { if ($si == SWITCH_IPADDRESS) { setflag(FLAG_FROM_SWITCH); } else { setflag(FLAG_FROM_USER); } } # 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()) { route(RELAY); } else { if (is_method("NOTIFY")) { route(RELAY); } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard #xlog("ACK without matching transaction ... ignore and discard"); exit; } } sl_send_reply("404","Not here"); } exit; } } onreply_route[EXTERNAL_REPLY] { route(NAT_TEST_AND_CORRECT); } route[NAT_TEST_AND_CORRECT] { if (nat_uac_test("3")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { #setbflag(FLB_NATB); route(RTPPROXY); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif route(RTPPROXY); /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; # if (!t_relay()) { # sl_reply_error(); # } # exit; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { #unforce_rtp_proxy(); force_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; }

2 1

RTPEngine wow's
by Wilkins, Steve 22 Aug '18

22 Aug '18

Hello all, I am still trying to get RTPEngine to work. At this point, when I make a call, I do see the offer in the rtpengine log, I also see the update of the SDP (c) changing it to 20.20.20.20. However, after that, there are no logs other than a bunch of "timer run time = 0.nnnnnnn sec", and the delete when I hang up the call. There is no RTP traffic what so ever in the rtpengine log (however, the call is completely successful, other than the RTP traffic not going through RTPEngine). Here is a breakdown of my configs (I borrowed this format from Maxim Fedorov because I thought it was very concise) Starting rtpengine => rtpengine --interface=internal/21.21.21.21--interface=external/21.21.21.21\!20.20.20.20 --listen-ng=127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1 RTPEngine Public IP: 20.20.20.20 RTPEngine Private IP: 21.21.21.21 kamailio.cfg modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12221") listen=tcp: 21.21.21.21:5060 advertise 20.20.20.20:5060 listen=udp: 21.21.21.21:5060 advertise 20.20.20.20:5060 rtpengine_manage call rtpengine_manage("trust-address replace-origin replace-session-connection external external media-address=20.20.20.20"); (I have placed this rtpengine_manage call at the beginning of all routes). I have tried many variations of rtpengine_manage parameters, and placement of the call itself, but I never see RTP traffic in the rtpengine log. Thank you to all who have looked at this, -Steve

1 0

Error at startup
by Nicolas Breuer 22 Aug '18

22 Aug '18

Hello, How can i clear this error ? Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:228]: pv_table_add(): pvar [rtpstat] already exists Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:513]: register_pvars_mod(): failed to register pseudo-variable <rtpstat> for module rtpproxy

2 2

Releasing Kamailio v5.1.5
by Daniel-Constantin Mierla 22 Aug '18

22 Aug '18

Hello, with the summer holidays in the northern hemisphere approaching the end, I am considering to release v5.1.5 out of Kamailio git branch 5.1 -- there were several fixes done since 5.1.4 and it is time to package another minor release. The propose date is next week on Wednesday, August 22, or slightly after in case there are unexpected events delays it. Should there be any issues you are aware of and not yet reported on the bug tracker, do it asap in order to have a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

1 2

Encoding mechanism in UAC Module
by Duarte Rocha 21 Aug '18

21 Aug '18

Greetings, Is there a way to choose the encoding algorithm that builds the vsf and vst parameters in the Record-Route? What is the default method that Kamailio uses for this ? Thanks in advance, Duarte Rocha

2 1

invalid version 8 for table aliases found, expected 9
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Hi everyone, i am trying to start kamailio 5.1.4 after the migration from Kamailio 3.0.3 but getting newly the following error: ERROR: <core> [db.c:450]: db_check_table_version(): invalid version 8 for table aliases found, expected 9 (check table structure and table "version") Allegedlly, this error is because of a wrong version of "aliases" table. I followed the Upgrade process starting from 3.0.3 till 5.1.x and I altered the structure of my aliase table accordinglly. The last new version of aliases is 8 as it is described at the upgrade from 4.2.x to 4.3.0 Even though, the error here says that the newest version is 9: May someone help me to find out if the aliases table's structure was renewed in between ?? Is there any modification in the data types? Thanks in advance Abdulaziz

3 4

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Greetings, i am at the final stages of emigrating from kamailio 3.0.3 to 5.1.0 including our own module. Unluckily, after writing the command " kamctl start", the following error is popping up: * PID file /var/run/kamailio.pid does not exist -- Kamailio start failed* Thus, I used the command " kamailio -M 8 -E -e -dd " to get more debugging Info and the output is detailing the following: 0(10334) ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on '192.168.1.207' (4) 0(10334) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(10334) ERROR: ABDUL_functions [ABDUL_chk.c:1069]: init_db(): failed to connect database URL=mysql://ABDUL:ABDUL@192.168.1.207/ABDUL 0(10334) ERROR: ABDUL_functions [ABDUL_functions_mod.c:296]: mod_init(): Module ABDUL_functions not correktly initialized, pid[10334] 0(10334) ERROR: <core> [core/sr_module.c:991]: init_mod(): Error while initializing module ABDUL_functions (//lib64/kamailio/modules/ABDUL_functions.so) Seemingly the error relates to old databases url ( "URL=mysql:// ABDUL:ABDUL@192.168.1.207/ABDUL" ) inside an old Kamailio.cfg file from my 3.0.3 package which I put under another different directory. This old Kamailio.cfg file #includes my own module, so he is exporting this url. Consequently, my module is trying to connect again to this databases. I do not know why both kamailio.cfg (new: /etc/kamailio/kamailio.cfg and old one out of /etc/kamailio) are both seen for kamctl or (/sbin/kamailio). I know that my new url inside the new /etc/kamailio/kamailio.cfg is read and connected but again not sure why my hidden old kamailio.cfg is still being read as well. the attached shows my init file for "SUSE Linux Enterprise Server 11 (x86_64)" and my used kamailio *default *file under /etc/default. Thanks in advance and best regards Abdulaziz

2 2

help
by Pravin . 21 Aug '18

21 Aug '18

Hello Team, I want to install Kamailio SIP server on centOS server...do we need to download kamailio ISO file and install on server..pls provide the installation procedure/guidelines. Regards Pravin

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