- sr-users - kamailio.org

kamctl ul command seems NOK when using an IPv6 Address
by Albert Petit 09 Jul '19

09 Jul '19

Hi, I am trying to create an identity with permanent location in my system using kamctl ul add command. Usually i have no issue doing that in IPv4 ,example kamctl ul add mcpttcontrolling sip:mcpttcontrolling@127.0.0.1:5070 it is OK However when i do that using an IPv6 address i get following error: kamctl ul add mcpttcontrolling sip:mcpttcontrolling@ [2001:470:1f21:219:20c:29ff:fedb:db6e]:5070 error: invalid SIP URI: sip:mcpttcontrolling@ [2001:470:1f21:219:20c:29ff:fedb:db6e]:5070 Let me know what you would need to investigate or what i am doing wrong for adding a location with IPv6 Address using kamctl tool Thanks Albert

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Jansson_get issues
by Igor Olhovskiy 09 Jul '19

09 Jul '19

Hi! I'm trying to get jannson_get working, but seems missing something. Example is quite simple, actually taken from wiki $var(result) = $null; jansson_get("pi", '{"mystr":"my input string", "count":9000, "mybool":true, "pi":3.14159}', $var(result)); When running this part of code getting ERROR: <core> [core/action.c:1139]: do_action(): runtime fixup failed for jansson_get param 3 ERROR: <core> [core/action.c:1518]: do_action(): run action error at: /etc/kamailio/kamailio.cfg:1030 Also tried with jannson_get(".pi", ...); jannson_get('/"pi"', ...); and a bit others. What am I missing? kamailio 5.2.3 Thanks!

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KW 2019 presentation files
by Denys Pozniak 09 Jul '19

09 Jul '19

Hello! Where can I find fresh presentation files from the KW 2019? PS. Nothing here: https://www.kamailio.org/events/ -- BR, Denys Pozniak

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where can i find siremis user guide?
by quxiao＠unioncast.cn 09 Jul '19

09 Jul '19

hi�� i have installed siremis successfully for the first time, but there is no guide to use siremis, where can i find it, thanks Best Regards & Thanks�� ֻ��13401129519 ��䣺quxiao(a)unioncast.cn ְλ��߼�C++��ʦ ��ţ��з�� Vҕͨ�_��־W�j�Ƽ��޹�˾ Unioncast Digital Network Technology Co., Ltd. Beijing ��˾��ҳ��http://www.unioncast.cn/ һ��ƶ��Ӧ�÷��ṩ�� ַ��к��йش��ϴ��3�ź��Ƽ��9��910�� ʱࣺ100081

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Kamailio new in devel for upcoming 5.3.0
by Henning Westerholt 08 Jul '19

08 Jul '19

Hello, I have updated the "new in devel" features document in our wiki. Pretty amazing stuff already in for the upcoming 5.3.0: * 5 new modules * Changes in 32 existing modules * Many changes in core, API refactoring and tools * About 1000 commits from 62 authors (development still ongoing!) Full list at: https://www.kamailio.org/wiki/features/new-in-devel A summary can be found in my today talk at COMMCON, slides 10,11: https://skalatan.de/de/archiv/presentations/commcon-2019-presentation.pdf Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

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How to statelessly send a reply to custom destination?
by George Diamantopoulos 05 Jul '19

05 Jul '19

Hello all, I've been trying to handle OPTIONS statelessly, but proper handling fails for UACs behind NAT without ALG. I thought I'd overcome this by setting the $du pseudovariable, but apparently I'm missing something. The following does not work: if ($Rp == "6050") { force_send_socket(udp:10.10.10.10:6050); } $du = "sip:" + $si + ":" + $sp; sl_send_reply("200","OK"); exit; I'm now realising that I put this in request_route() so force_send_socket() and $du have no use whatsoever, since the request is not forwarded anywhere. Is this the reason? So the question is, how can I do the following things for a stateless reply: a) choose a sending socket for the reply and b) change the destination for the reply to something other than the URI in the first Via header. Thanks! BR, George

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Append a new branch while the other one is in ringing state
by Koray Vatansever 04 Jul '19

04 Jul '19

Hi all, I want to fork a new branch, while the first one is ringing. Forking will take place in a rtimer process. So I will save the invite transaction in a htable, and using this transaction info I need to append a branch. I saw TSILO module provides some methods to append branch. However documentation says that it can be used when register message received. What is the best way to implement this? Thanks, Koray

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people complain Kamailio not handling stale nonce correctly
by Juha Heinanen 03 Jul '19

03 Jul '19

Several people are complaining on baresip git that Kamailio is among those SIP servers that don't handle stale nonce properly, i.e., don't include stale=true flag in WWW-Authenticate header of 401 response. I have tried to tell that Kamailio does what it has been configured to do, but the response is that "by default" Kamailio works incorrectly. "By default" I guess refers to the example Kamailio kamailio.cfg in etc dir. How about adding stale nonce handling to the example config, for example, like in below? -- Juha *** kamailio.cfg 2019-06-13 17:17:17.182148989 +0300 --- /tmp/kamailio.cfg 2019-07-02 18:02:02.947183503 +0300 *************** *** 745,753 **** 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")) --- 745,759 ---- if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ! switch (auth_check("$fd", "subscriber", "1")) { ! case 1: ! break; ! case -4: ! auth_challenge("$fd", 17); ! exit; ! default: ! auth_challenge("$fd", "0"); ! exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH"))

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Change
by Samir Kundu 03 Jul '19

03 Jul '19

Please note change of my new email address – samirkundu(a)samirkundultd.plus.com Thank you Samir ====================================================================== COMPUTING & COMMUNICATION HARDWARE & SOFTWARE SOLUTIONS PROVIDER ====================================================================== e-mail is confidential --- "SAVE PAPER - THINK BEFORE YOU PRINT!" Samir Kundu Ltd | Tel: 0191 538 8280 | Email: samir_kundu(a)samirkundugroup.co.uk Address: 3 Beechwood Terrace, Sunderland SR2 7LY Registered in England & Wales No. 09051111 ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

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DMQ: dmq_t_replicate()
by Lars Olsson 02 Jul '19

02 Jul '19

Hi, I am trying to replicate a REGISTER request between several nodes. When I am using dmq_t_replicate(), it ends the execution of the script. It this correct or I have missed anything? I can not find anything in the documentation of the DMQ module the says that it will end the script execution. Is it recommended to use the DMQ module or should I simply forward the request instead as described in https://medium.com/@tumalevich/kamailio-registration-replication-without-dm… Best Regards, Lars

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[DMQ] Sync DLG_STATE_CONFIRMED_NA dialog state ?
by Mathieu Bodjikian 02 Jul '19

02 Jul '19

Hello, We are using DMQ module to sync dialogs between 3 Kamailio boxes. We are using last Kamailio version (5.2.3). >From our understanding, the state DLG_STATE_CONFIRMED_NA is not synced over DMQ (https://github.com/kamailio/kamailio/blob/a84a3ea618f0e602a8892c37fce7f4e72…) With containerized environement, the ACK's can go to any Kamailio instance, which lead to : On first box : -> INVITE received -> Dialog created + KDMQ sent (state DLG_STATE_UNCONFIRMED) -> 180 | 183 received -> Dialog updated + KDMQ sent (state DLG_STATE_EARLY) -> 200 received -> Dialog updated + KDMQ NOT sent (DLG_STATE_CONFIRMED_NA) On second box : -> ACK received : - next_state_dlg(): bogus event 6 in state 2 for dlg - dialog not updated - KDMQ with state DLG_STATE_CONFIRMED not sent On first box, after timeout : -> After timeout : - dialg with incorrect timeouts (since box2 didn't send dialog update) - tm sends BYE in both ways - bye_reply_cb(): inconsitent dlg timer data on dlg We see two solutions : - We add DLG_STATE_CONFIRMED_NA to states we sync on dlg_dmq_replicate_action - We manage to route the ACK on the same instance that received first INVITE (could be tricky) Which one is preferable ? Kind regards, Mathieu Bodjikian

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question about reply route
by Iván Aponte 02 Jul '19

02 Jul '19

Hello, I have a question about record route and via headers. My scenario is the following: [asterisk] ---- [kam] -------- [trunk] 10.142.0.27 10.142.0.6 200.x.x.x I have an asterisk an a kamailio in a private ip network (10.142.0.0/24). Kamailio has the address 10.142.0.6 but is also natted 1:1 to a public ip (35.0.0.6). My problem is that when the invite are replied the asterisk replies to the external ip (35.0.0.6) instead of the internal ip (even though the kamailio replies goes through the internal network) . I tried adding a reply route to change the advertised address to no avail. any advice would be appreciated. this is my conf: LISTEN="listen=udp:0.0.0.0:5060 advertise 35.0.0.6:5060" disable_tcp=yes loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "permissions.so" loadmodule "ipops.so" modparam("jsonrpcs", "pretty_format", 1) modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 0) reply_route { xinfo("source $si"); if(!is_in_subnet($si, "10.0.0.0/8")) { set_advertised_address("10.142.0.6"); } else { xinfo("not in subnet"); } } request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication #route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # dispatch requests to foreign domains route(SIPOUT); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } } # Wrapper for relaying requests route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { send_reply("405", "Method Not Allowed"); 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 route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } -- Iván Aponte Office: +58(212)9923193 Mobile: +58(412)2774713

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Re: [SR-Users] sr-users Digest, Vol 170, Issue 1
by ali mahfouz 02 Jul '19

02 Jul '19

chat over sip where and how save it On Mon, Jul 1, 2019 at 1:03 PM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: ATCF/ATGW functionality (Mojtaba) > 2. Re: ATCF/ATGW functionality (Mojtaba) > 3. Re: ATCF/ATGW functionality (Maryam Baghdadi) > 4. kamailio wiht PJSIP IOS client (Gaurav Bmotra) > 5. Re: ATCF/ATGW functionality (Mojtaba) > 6. Re: Late offer ACKs not passing through branch_route > (Enrico Bandiera) > 7. kamailio 4.4.4 fills IPv6 contact address when sending packet > over IPv4 (Pali Rohár) > 8. Re: install mongoDB module with kamailio-cli doker image > (Gaurav Bmotra) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 30 Jun 2019 15:40:07 +0430 > From: Mojtaba <mespio(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Cc: users(a)lists.kamailio.org > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > < > CABVi_Ew5Ti0bH5z4kcoBwx9jvv8S06GtEqgzCKPfxE7jfh7aJQ(a)mail.gmail.com> > Content-Type: text/plain; charset="UTF-8" > > Unfortunately you can't, because SRVCC is a capable between MME and > MCS in your network. When the UE is moving on, the MME send "Take > handover of the call" to MSC and move the surrect session control to > MSC. You could read more about SRVCC in 3GPP TS 29.280. > And FHOSS does not support interface between MME and HSS, > With Best Regards.Mojtaba > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > <maryam.baghdadi(a)gmail.com> wrote: > > > > Hello, > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > --Mojtaba Esfandiari.S > > > > ------------------------------ > > Message: 2 > Date: Sun, 30 Jun 2019 15:40:07 +0430 > From: Mojtaba <mespio(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Cc: users(a)lists.kamailio.org > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > < > CABVi_Ew5Ti0bH5z4kcoBwx9jvv8S06GtEqgzCKPfxE7jfh7aJQ(a)mail.gmail.com> > Content-Type: text/plain; charset="UTF-8" > > Unfortunately you can't, because SRVCC is a capable between MME and > MCS in your network. When the UE is moving on, the MME send "Take > handover of the call" to MSC and move the surrect session control to > MSC. You could read more about SRVCC in 3GPP TS 29.280. > And FHOSS does not support interface between MME and HSS, > With Best Regards.Mojtaba > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > <maryam.baghdadi(a)gmail.com> wrote: > > > > Hello, > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > --Mojtaba Esfandiari.S > > > > ------------------------------ > > Message: 3 > Date: Sun, 30 Jun 2019 16:32:01 +0430 > From: Maryam Baghdadi <maryam.baghdadi(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > <CAPrdAbAnnTULTA1rZ-b= > PG1FE+4HfqRHtKg_3R49WEBUXbM+Fw(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Thank you for your response, > If we need to handover of call in our network, What's the solution? is it > possible to add external MSC Server to out network? or add a SBC with this > ability ? > > On Sun, Jun 30, 2019 at 3:40 PM Mojtaba <mespio(a)gmail.com> wrote: > > > Unfortunately you can't, because SRVCC is a capable between MME and > > MCS in your network. When the UE is moving on, the MME send "Take > > handover of the call" to MSC and move the surrect session control to > > MSC. You could read more about SRVCC in 3GPP TS 29.280. > > And FHOSS does not support interface between MME and HSS, > > With Best Regards.Mojtaba > > > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > > <maryam.baghdadi(a)gmail.com> wrote: > > > > > > Hello, > > > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > > > _______________________________________________ > > > Kamailio (SER) - Users Mailing List > > > sr-users(a)lists.kamailio.org > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > --Mojtaba Esfandiari.S > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > >

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kamailio wiht PJSIP IOS client
by Gaurav Bmotra 01 Jul '19

01 Jul '19

hi i na using kamailio 5.1.2 and i m developing ios client with PJSIP audio is working fine ..but on video it work one side but not showing other side it is working fine on other client like zoiper for audio/video both can anyone help me -- *Regards:* Gaurav Kumar

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IMS strange behaviour
by Amar Tinawi 01 Jul '19

01 Jul '19

Hello list i'm facing strange behavior in my lap which consists of p-cscf, i-cscf, s-cscsf, FHoSS for HSS Registration is ok and ICSCF is performing well but when it comes to calls, INVITE Call between two ims users is not passing from scscf to icscf the flow should be : P-->S-->I-->hss i can see invite massage is going out scscf to icscf with all monitoring tools, but no incoming Invite massage to icscf not on sip level and nor on networking level (tcpdump) Actually i don't know where the INVITE packets are going Any advice Regards

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install mongoDB module with kamailio-cli doker image
by Gaurav Bmotra 01 Jul '19

01 Jul '19

hi i wan to install mongodb module in kamailio/kamailio-ci image pl help me thnaks -- *Regards:* Gaurav Kumar

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Late offer ACKs not passing through branch_route
by Bandiera, Enrico 01 Jul '19

01 Jul '19

Hi, I'm encountering an issue when relaying late offer ACKs, before relaying I set the branch_route with t_on_branch() but kamailio never enters that route after calling t_relay I'm keeping the rtpengine module calls inside branch_routes to avoid issues when re-routing in failure routes I suppose it's something related to ACK packets but I'm not sure what the issue is. Any hints? Thanks, Enrico.

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ATCF/ATGW functionality
by Maryam Baghdadi 01 Jul '19

01 Jul '19

Hello, Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks.

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Kamailio 5.x and Call_control module
by Efelin Novak 30 Jun '19

30 Jun '19

Hi Folks, I'm migrating to Kamailio 5.2.x from 4.4.x. Everything seems to be fine, however I have came to an issue with call_control module as this one is still using old MI interface. Standard situations work nice (maximum debit, prepaid, "CDRs") however when call_control needs to kill a call (credit is gone), it tries to send dlg_end_dlg over MI and it fails. Question are: Is call_control still supported? I haven't found any note about it. Is there any workaround from Kamailio point of view? I'm running last versions of both applications. I'm also rising a ticket at ag-projects side. Thanks Efelin

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Re: [SR-Users] sr-users Digest, Vol 169, Issue 27
by ali mahfouz 29 Jun '19

29 Jun '19

if I send my kamailio.cfg file can you edit it to me please because it is so urgent On Thu, Jun 27, 2019 at 1:06 PM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: TOPOS uid in contact param instead of username > (Daniel-Constantin Mierla) > 2. Kamailio 5.x and Call_control module (Efelin Novak) > 3. Re: Kamailio 5.x and Call_control module > (Daniel-Constantin Mierla) > 4. Statistics for locally generated replies (Duarte Rocha) > 5. Re: Kamailio 5.x and Call_control module > (Daniel-Constantin Mierla) > 6. Re: Kamailio 5.x and Call_control module (Efelin Novak) > 7. Re: kamailio as UAC and NAPTR / sendsocket question > (Karsten Horsmann) > 8. MongoDB for Kamailio (Gaurav Bmotra) > 9. Presence publish manipulation (Gertjan Wolzak) > 10. Re: Question about registrar behavior > (Володимир Іванець) > 11. Re: MongoDB for Kamailio (Henning Westerholt) > 12. Re: DMQ: dmq_t_replicate() (Henning Westerholt) > 13. Re: does kamailio support graph database ?? (Henning Westerholt) > 14. Re: Question about registrar behavior (Henning Westerholt) > 15. Need help with "no branches for forwarding" message (Andrew Chen) > 16. Topos / Remove HF_re (Nicolas Breuer) > 17. Re: Kamailio 5.x and Call_control module (Efelin Novak) > 18. Re: Question about registrar behavior > (Володимир Іванець) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Jun 2019 12:52:05 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] TOPOS uid in contact param instead of username > Message-ID: > < > CAFRry4VMJd9aaJ_teWd3E-CYg-O9K+qutW699n9GDc9is7wAxQ(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hello, > ok, feel free to ask question if something is not clear ... and I think > topoh might help looking at its code, iirc, it uses uri param for contact > masking ... > > Anyhow, there is no better fun that C coding ;-) ... > Cheers, > Daniel > > On Wed, Jun 26, 2019 at 10:12 AM Thomas Weber <thomas.weber(a)pascom.net> > wrote: > > > Hello, > > > > thank you for your opinions. > > > > So i will give it a try throughout the next weeks because we have some > > customer pressure here. > > Didn't do a C project since some 10 years, so it could be fun to "learn > it > > again". > > > > Cheers, > > > > > > Thomas > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >

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Re: [SR-Users] Kamailio next_state_dlg(): bogus event
by Olli Attila 29 Jun '19

29 Jun '19

Hello, The prack is for the initial INVITE establishing the call. No re-INVITE messages are generated in the dialog. --Olli Message: 11 Date: Fri, 28 Jun 2019 08:35:09 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamailio next_state_dlg(): bogus event Message-ID: <CAFRry4XAF5qYqRxgJA4tzsuUt-56ON1KnzzwvJnxeKOdTUaxSA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hello, is the PRACK following a re-INVITE or is for the initial INVITE establishing the call? Cheers, Daniel On Fri, Jun 28, 2019 at 8:13 AM Olli Attila <attiolli(a)gmail.com> wrote: > Hello, > > I'm getting this error in Kamailio log every now and then. Can anyone > suggest methods to debug this further for root cause? > > Jun 27 10:04:09 /usr/sbin/kamailio[26341]: CRITICAL: dialog > [dlg_hash.c:1098]: next_state_dlg(): bogus event 5 in state 4 for dlg > 0x7fceeabf8d38 [749:3185] with clid > 'SDceked01-e4d2e93d7f9ffe88907206a25ccdbcc2-v300g00030' and tags > 'SDceked01-ecd4281b' '14001358846426' > > Behaviour in the call: > When I look at the SIP flow from this call I can see that a PRACK > message which is routed to Kamailio gets discarded (=the proxy does > not route the PRACK onwards to the UA that requested the usage of the > 100rel in 183 session progress message.) The PRACK message itself > seems valid and when compared to a PRACK from a working call, there is > no difference in the content. > > Cheers, > Olli

2 1

Htable not reloading from db on expiry
by Daniel Greenwald 28 Jun '19

28 Jun '19

I have an htable that loads from mysql. The data loads fine on startup but disappears on expiry. If I issue kamcmd htable.reload it loads again until expiry. It seems as if db_expires is not taking affect. I see them expiring in the debug log but not an errors. Mysql general log does not show any attempt to connect on expiry. Another odd thing is that if I set the expires column in db to anything other than 0 the data doesn't load at all. Any ideas would be appreciated. modparam("htable", "timer_interval", 10) modparam("htable", "htable", "h=>size=20;autoexpire=20;dbtable=validdomains;updateexpire=0;") modparam("htable", "db_url", AUTHDBURL) modparam("htable", "db_expires", 1) create or replace view validdomains as select distinct subscriber.domain as key_name, 1 as key_value, 0 as key_type, 0 as value_type, 0 as expires from subscriber;

2 1

Kamailio next_state_dlg(): bogus event
by Olli Attila 28 Jun '19

28 Jun '19

Hello, I'm getting this error in Kamailio log every now and then. Can anyone suggest methods to debug this further for root cause? Jun 27 10:04:09 /usr/sbin/kamailio[26341]: CRITICAL: dialog [dlg_hash.c:1098]: next_state_dlg(): bogus event 5 in state 4 for dlg 0x7fceeabf8d38 [749:3185] with clid 'SDceked01-e4d2e93d7f9ffe88907206a25ccdbcc2-v300g00030' and tags 'SDceked01-ecd4281b' '14001358846426' Behaviour in the call: When I look at the SIP flow from this call I can see that a PRACK message which is routed to Kamailio gets discarded (=the proxy does not route the PRACK onwards to the UA that requested the usage of the 100rel in 183 session progress message.) The PRACK message itself seems valid and when compared to a PRACK from a working call, there is no difference in the content. Cheers, Olli

2 1

Statistics for locally generated replies
by Duarte Rocha 27 Jun '19

27 Jun '19

Greetings, Does the statistics command "kamcmd tm.stats" has an argument or a way to only list locally generated replies? As far as i know, sl.stats only lists local generated replies. Can tm work the same way? Best Regards, Duarte Rocha

2 1

Need help with "no branches for forwarding" message
by Andrew Chen 27 Jun '19

27 Jun '19

Hey guys, I need help to identify the cause of this is error: Jun 26 12:03:53 ashintgtpsg51 /usr/sbin/kamailio[1651]: ERROR: tm [t_fwd.c:1728]: t_forward_nonack(): no branches for forwarding This message was shortly printed after I xlog a message in onsend_route[] block. What else is strange is that the dialog continued on with that same SIP Call-ID. This is what it looks like in the wireshark flow diagram: [image: image.png] The dispatcher entry is an SRV record which lists available hosts. Any thoughts on how I can identify the cause of this? 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.*

1 1

Topos / Remove HF_re
by Nicolas Breuer 27 Jun '19

27 Jun '19

Hello, When using TOPOS module, Kamailio add this header : P-SR-XBranch before executing the topos module. Doing this "remove_hf_re("P-")" in the scripting seems to bypass the addition of the P-SR-XBranch in the sip message. In my mind, remove header function just remove the headers based on the incoming message. What do you think ?

2 1

Question about registrar behavior
by Lars Olsson 27 Jun '19

27 Jun '19

Hi, I have found a behavior in the registrar module that I do have a question about. Is the current behavior correct and wanted? Using the save() method in the script I see the following: * Processing a register request for a user gives return code 1 ( or 2 ) * Processing a unregister request (expires=0) for registered user gives return code 3 * Processing a unregister request (expires=0) for a user which is NOT registered gives return code 1. Why? What is the reason behind this? No database entry is added which is expected. Test performed on 5.1.4, using DB mode 3. For handling a late unregister request ( where registration has already expired) return code does not reflect the the action. I assume that manually checking $expires(max) is the option to go then if I want to detect the unregister request or? Cheers, Lars

3 3

does kamailio support graph database ??
by Gaurav Bmotra 26 Jun '19

26 Jun '19

hi i m using kamailio 5.1 , does kamailio support graph database ?? thank you -- *Regards:* Gaurav Kumar

2 1

MongoDB for Kamailio
by Gaurav Bmotra 26 Jun '19

26 Jun '19

hi i m using kamailio 5.1 , on Ubuntu 18.4 and right now i m using Maria DB for kamailio , but i want to migrate my Maria data to MingoDB ,, plz help me how can i use MOngoDB with kamailio thanks -- *Regards:* Gaurav Kumar

2 1

Presence publish manipulation
by Gertjan Wolzak 26 Jun '19

26 Jun '19

Hello Kamailions, I'm running into the following challenge. A kamailio setup where phones register with their mac address as uid and a password. Users have to login to the system by dialing a special number where they have to enter their internal extension and a pin code. Now the system knows which extension(number) is logged in on which phone(macaddress). This is done with from updates, just a type of hotdesking. Now I want to use presence.... When a phone makes a phone call, kamailio generates a publish message where the "local" part of the xml code contains the mac address. How can I manipulate the Publish xml information? I mean the manipulation can be done with textops, but which variables to use and where in the configuration? Im asking because when I disable the presence route and do not call it in my configuration, but with presence enabled, I still get Publish messages.... I have tried adding a Sender header which contains the "extension" of the phone and then call the publish in the PRESENCE route like this: handle_publish("$hdr(Sender)"); But this does not change the xml information. Am I doing things wrong? Below the first publish message that is send, where 10.88.77.172 is the kamailio server. U 2019/06/26 14:54:37.233345 10.88.77.172:5060 -> 10.88.77.172:5060 #7 PUBLISH sip:0015659a9931@10.88.77.172:5060 SIP/2.0. Via: SIP/2.0/UDP 10.88.77.172;branch=z9hG4bK1a87.5510d093000000000000000000000000.0. To: <sip:0015659a9931@10.88.77.172:5060>. From: <sip:0015659a9931@10.88.77.172:5060>;tag=a2652bb825a097f3b7285d4c70edd51a-5c5d. CSeq: 10 PUBLISH. Call-ID: 173009c21f4e3c77-94480(a)10.88.77.172. Content-Length: 579. User-Agent: kamailio (5.1.8 (x86_64/linux)). Max-Forwards: 70. Event: dialog. Expires: 125. Content-Type: application/dialog-info+xml. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:0015659a9931@10.88.77.172:5060"> <dialog id="0_2285997339(a)10.88.77.184" call-id="0_2285997339(a)10.88.77.184" direction="initiator"> <state>Trying</state> <remote> <identity>sip:604113@10.88.77.172:5060</identity> <target uri="sip:604113@10.88.77.172:5060"/> </remote> <local> <identity>sip:0015659a9931@10.88.77.172:5060</identity> <target uri="sip:0015659a9931@10.88.77.172:5060"/> </local> </dialog> </dialog-info> I would like to change the local identity and target uri into the extension assigned to that phone, for example 604114.... So that subscribers to the extension 604114 get informed about the state of the phone with the used uid 0015659a9931. Feedback would be appricated. Pointers to where I am going wrong as well, ... that's feedback too. Rgds, Gertjan

1 0

TOPOS uid in contact param instead of username
by Thomas Weber 26 Jun '19

26 Jun '19

Hello, we're using topos/redis since some time and it's working quite well for most situations. It's used to route asterisk traffic to sip providers. Some providers, however, are validating the contact header strictly against i.e. the accounts username and so we have to disable topos by using event_route[topos:msg-sending]. Now i'm wondering if it would be useful to other people to add a mode to topos where the topos uid is not forwarded as contact username but instead added as a contact uri param. So instead of e.g. Contact: <sip:btpsh-5abcdef5-374-1@x.x.x.x:5060;transport=tcp> it would use Contact: <sip:+49......@x.x.x.x:5060;transport=tcp;line=btpsh-5abcdef5-374-1>. What do you think? Would it make sense? Is it a lot of effort? Cheers, Thomas

4 4

Apache version for FHoSS
by Woscek, Martin W. 26 Jun '19

26 Jun '19

Hello all, What is the version of Apache that FHoSS was built to? Thanks, Martin

2 1

Info: ClueCon Weekly - Kamailio Project Updates
by Daniel-Constantin Mierla 25 Jun '19

25 Jun '19

Hello, later today I will join ClueCon Weekly for a discussion about what is new for Kamailio project and the plans for the future. ClueCon Weekly is a web video conference organized by the FreeSwitch project team. They announced recently a revamp of the format, however, I expect that other people can still join live if they want to listen and ask questions, couple of ways to connect being listed at: * https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… If joining live is no longer possible, I will post a link to the recording and if someone has any questions following my presentation, I can answer here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 4

Carrier Route installation Failed
by Amar Tinawi 24 Jun '19

24 Jun '19

Hello i'm facing a dependency error when trying to install kamailio 5.2.3+xenial with carrier route module using apt package installer The following packages have unmet dependencies: kamailio-carrierroute-modules : Depends: kamailio (= 4.3.4-1.1ubuntu2) but 5.2.3+xenial is to be installed E: Unable to correct problems, you have held broken packages. all other modules installed fine any suggestions? thanks in advance

4 5

Kamailio routing BYE back to originator
by Andrew White 24 Jun '19

24 Jun '19

Hi team! I’ve recently found an issue in my Kamailio setup. It appears when the trunk replies to call in progress with a BYE, we relay it immediately back to them: https://i.imgur.com/h5fusau.png <https://i.imgur.com/h5fusau.png> The only differences in the second BYE is the Route header is removed and replaced with a Via header referencing the Kamailio machine, and the From URI is rewritten to show the Kamailio hostname. I’m unsure what steps I can take here, as all of my out of dialog messages (180, 200, ACK) are forwarded along with no issue. It seems only the in dialog are having this issue. Here’s my WITHINDLG and RELAY: def ksr_route_relay() if KSR.is_method_in("IBSU") then if KSR::TM.t_is_set("branch_route") < 0 then KSR::TM.t_on_branch("ksr_branch_manage") end end if KSR.is_method_in("ISU") then if KSR::TM.t_is_set("onreply_route") < 0 then KSR::TM.t_on_reply("ksr_onreply_manage") end #KSR.info("Onreply route: #{KSR::TM.t_is_set("onreply_route")}") end if KSR.is_INVITE() then if KSR::TM.t_is_set("failure_route") < 0 then KSR::TM.t_on_failure("ksr_failure_manage") end end if KSR::TM.t_relay() < 0 then KSR::SL.sl_reply_error() end exit end def ksr_route_withindlg() return if KSR::SIPUTILS.has_totag() < 0 if KSR::RR.loose_route() > 0 then ksr_route_dlguri() if KSR.is_BYE() then #KSR.setflag($myenv['FLT_ACC'].to_i) #KSR.setflag($myenv['FLT_ACCFAILED'].to_i) elsif KSR.is_ACK() then ksr_route_natmanage() elsif KSR.is_NOTIFY() then KSR::RR.record_route() end ksr_route_relay() exit end if KSR.is_ACK() then #KSR.info("Handling an ACK within dialog") if KSR::TM.t_check_trans() > 0 then ksr_route_relay() exit else exit end end #KSR.info("404ing within dialog") KSR::SL.sl_send_reply(404, "Not here"); exit end I suspect I must be sending this to the wrong branch somehow, but I’m unsure how to correct this. The call gets originally initiated from Kamailio writing a custom RURI and To header before calling t_relay(). I’m wondering if I’m somehow initiating the dialog incorrectly and as such the BYE doesn’t know where to go. Thanks! Andrew

2 3

Kamailio config file flow chart
by Gaurav Bmotra 22 Jun '19

22 Jun '19

hi i m looking for kamailio config file flow chart ,,, if any one have plz provide thanks -- *Regards:* Gaurav Kumar

2 1

5.2.3 - dispatcher module looses setid
by Andrew Chen 21 Jun '19

21 Jun '19

Hi all, For a few times now I've noticed Kamailio 5.2.3 dispatcher module looses view of setid destination and this system been up and running for 13 days: root:~# kamcmd dispatcher.list | grep ash.tpgw.service To work around it, we have to reload the module again: root:~# kamcmd dispatcher.reload root:~# kamcmd dispatcher.list | grep ash.tpgw.service URI: sip:<some host>:5062 URI: sip:<some host>:5060 root:~# Process been running for several days: root:~# ps -elf | grep kam 0 S root 8484 6520 0 80 0 - 3714 pipe_w 16:04 pts/0 00:00:00 grep --color=auto kam 5 S kamailio 32086 1 0 80 0 - 214845 pause Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32164 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32165 32086 0 80 0 - 233342 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32166 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32167 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32168 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32169 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32170 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio Anyone seen this? Thanks -- Andy Chen Sr. Telephony Lead Engineer 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 3

Kamailio doesn't start with TLS
by Олег Подгуйко 21 Jun '19

21 Jun '19

Im trying to use kamailio with websocket and tls. It will be use as registrar for websocket clients. I'm using default config with additional websocket, https and tls modules. I created some certs for it and everething looks like good. But when I started kamailio via (systemctl start kamailio) I get some errors июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) DEBUG: <core> [core/tcp_main.c:2753]: tcp_init(): added 37.143.9.210:443 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR: <core> [core/tcp_main.c:2855]: tcp_init(): bind(9, 0x7fb6ef026f84, 16) on 37.143.9.210:443 : Permission denied июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR: tls [tls_init.c:342]: tls_h_init_si(): Error while initializing TCP part of TLS socket 37.143.9.210:443 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) DEBUG: rtpengine [rtpengine.c:1986]: mod_destroy(): rtpengine_hash_table_destroy() success! If I start it via /usr/sbin/kamailio - all work. What I do wrong? -- Олег Подгуйко

5 8

kamcmd uac.* commands dont find records
by Karsten Horsmann 21 Jun '19

21 Jun '19

Good morning List! I have an issue with my kamcmd from kamailio 5.2.3 on CentOS 7.6 with kamailio-rpms. The kamcmd util can not find my registrations via command kamcmd uac.reg_enable auth_username FFFFFFFFFFFF kamcmd uac.reg_info l_username +49XXXXXXXX and soone. Only global commands for uac works. Whats going wrong with my kamcmd? Any ideas? Thank for your hints! # kamcmd uac.reg_dump { l_uuid: 49XXXXXXX l_username: +49XXXXXXXX l_domain: sip-trunk.telekom.de r_username: +49XXXXXXXX r_domain: sip-trunk.telekom.de realm: sip-trunk.telekom.de auth_username: FFFFFFFFFFFF auth_password: *************** auth_proxy: sip:reg.sip-trunk.telekom.de expires: 360 flags: 20 diff_expires: 129 timer_expires: 1561103682 reg_init: 1561101919 reg_delay: 3 } # kamcmd uac.reg_enable auth_username FFFFFFFFFFFF error: 404 - Record not found -- Kind regards Mit freundlichen Grüßen *Karsten Horsmann*

2 4

Re: [SR-Users] FHoSS db web interface failing for retrievals -UPDATE
by Woscek, Martin W. 20 Jun '19

20 Jun '19

I can only compile down to JDK 1.6 java 6 but still get same error below from the web interface. But get the following error when pointing to JDK1.5 java 5 which would imply some parts of the code was built with 1.6. BTW I downgraded Ant to 1.9 (Apache Ant(TM) version 1.9.14 compiled on March 12 2019). Buildfile: /opt/OpenIMSCore/FHoSS/build.xml init: [mkdir] Created dir: /opt/OpenIMSCore/FHoSS/bin compile: [javac] /opt/OpenIMSCore/FHoSS/build.xml:48: warning: 'includeantruntime' was not set, defaulting to build.sysclasspath=last; set to false for repeatable builds [javac] Compiling 202 source files to /opt/OpenIMSCore/FHoSS/bin [javac] /opt/OpenIMSCore/FHoSS/src/de/fhg/fokus/hss/auth/AuthenticationVector.java:49: cannot access de.fhg.fokus.diameter.DiameterPeer.data.AVP [javac] bad class file: /opt/OpenIMSCore/FHoSS/lib/jdp.jar(de/fhg/fokus/diameter/DiameterPeer/data/AVP.class) [javac] class file has wrong version 51.0, should be 49.0 [javac] Please remove or make sure it appears in the correct subdirectory of the classpath. [javac] import de.fhg.fokus.diameter.DiameterPeer.data.AVP; [javac] ^ [javac] 1 error BUILD FAILED /opt/OpenIMSCore/FHoSS/build.xml:48: Compile failed; see the compiler error output for details. Any insight would be appreciated. Thanks, Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Woscek, Martin W. Sent: Tuesday, June 18, 2019 2:38 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] FHoSS db web interface failing for retrievals Hi all, We can enter user data into our HSS, we just cant retrieve data. Just to verify, FHoSS is based on Java 1.7 and java.lang.CharSequence is in Java 1.7? What we have: java version "1.7.0_80" Java(TM) SE Runtime Environment (build 1.7.0_80-b15) Java HotSpot(TM) 64-Bit Server VM (build 24.80-b11, mixed mode) The error: HTTP Status 500 - ________________________________ type Exception report message description The server encountered an internal error () that prevented it from fulfilling this request. exception org.apache.jasper.JasperException: Unable to compile class for JSP An error occurred at line: 116 in the jsp file: /pages/user/imsu_search_result.jsp Generated servlet error: The type java.lang.CharSequence cannot be resolved. It is indirectly referenced from required .class files org.apache.jasper.compiler.DefaultErrorHandler.javacError(DefaultErrorHandler.java:84) org.apache.jasper.compiler.ErrorDispatcher.javacError(ErrorDispatcher.java:328) org.apache.jasper.compiler.JDTCompiler.generateClass(JDTCompiler.java:397) org.apache.jasper.compiler.Compiler.compile(Compiler.java:288) org.apache.jasper.compiler.Compiler.compile(Compiler.java:267) org.apache.jasper.compiler.Compiler.compile(Compiler.java:255) org.apache.jasper.JspCompilationContext.compile(JspCompilationContext.java:556) org.apache.jasper.servlet.JspServletWrapper.service(JspServletWrapper.java:293) org.apache.jasper.servlet.JspServlet.serviceJspFile(JspServlet.java:291) org.apache.jasper.servlet.JspServlet.service(JspServlet.java:241) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) org.apache.struts.action.RequestProcessor.doForward(RequestProcessor.java:1063) org.apache.struts.tiles.TilesRequestProcessor.doForward(TilesRequestProcessor.java:263) org.apache.struts.action.RequestProcessor.processForwardConfig(RequestProcessor.java:386) org.apache.struts.tiles.TilesRequestProcessor.processForwardConfig(TilesRequestProcessor.java:318) org.apache.struts.action.RequestProcessor.process(RequestProcessor.java:229) org.apache.struts.action.ActionServlet.process(ActionServlet.java:1194) org.apache.struts.action.ActionServlet.doPost(ActionServlet.java:432) javax.servlet.http.HttpServlet.service(HttpServlet.java:709) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) Thanks, _Martin

1 0

XMPP
by David Villasmil 19 Jun '19

19 Jun '19

Hello guys, I'm kind of confused as to what module exactly i need to user, let me explain: I have a service for DIDs and SMS from s.ms, i created a python script that connects to the service using xmpp to their jabber server with a user and a password, i.e.: xmpp_client.py MY-NUMBER(a)s.ms MY-PASSWORD What i do with that script is basically connect and wait to receive SMS, which i then insert in the proxy db (silo) and they get delivered by msilo. I have several numbers and doing this for every number seems stupid, so i started looking for a better solution, since the end user will be receive the message via SIP MESSAGE on they sip client. i came across kamailio/opensips' jabber and xmpp modules, but i'm not sure which (if any) to use, and even HOW to, or the actual architecture, since the documentation keeps talking about a local xmpp server? I would REALLY appreciate some help here. Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 2

Question on Database Version Table
by JR Richardson 18 Jun '19

18 Jun '19

Hey All, I'm doing some upgrades and noticed database table version numbers have increased. database table adjustments: | address | production ver 3 | upgrade 5.2 ver 6 | need to add accountcode and notes column | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add ruri_pattern and priority column I run mysql master/slave so I'm concerned about the master database not having the correct version numbers or can I just update the version number in the table when I do the upgrade of the kamailio versions? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

3 2

FHoSS db web interface failing for retrievals
by Woscek, Martin W. 18 Jun '19

18 Jun '19

Hi all, We can enter user data into our HSS, we just cant retrieve data. Just to verify, FHoSS is based on Java 1.7 and java.lang.CharSequence is in Java 1.7? What we have: java version "1.7.0_80" Java(TM) SE Runtime Environment (build 1.7.0_80-b15) Java HotSpot(TM) 64-Bit Server VM (build 24.80-b11, mixed mode) The error: HTTP Status 500 - ________________________________ type Exception report message description The server encountered an internal error () that prevented it from fulfilling this request. exception org.apache.jasper.JasperException: Unable to compile class for JSP An error occurred at line: 116 in the jsp file: /pages/user/imsu_search_result.jsp Generated servlet error: The type java.lang.CharSequence cannot be resolved. It is indirectly referenced from required .class files org.apache.jasper.compiler.DefaultErrorHandler.javacError(DefaultErrorHandler.java:84) org.apache.jasper.compiler.ErrorDispatcher.javacError(ErrorDispatcher.java:328) org.apache.jasper.compiler.JDTCompiler.generateClass(JDTCompiler.java:397) org.apache.jasper.compiler.Compiler.compile(Compiler.java:288) org.apache.jasper.compiler.Compiler.compile(Compiler.java:267) org.apache.jasper.compiler.Compiler.compile(Compiler.java:255) org.apache.jasper.JspCompilationContext.compile(JspCompilationContext.java:556) org.apache.jasper.servlet.JspServletWrapper.service(JspServletWrapper.java:293) org.apache.jasper.servlet.JspServlet.serviceJspFile(JspServlet.java:291) org.apache.jasper.servlet.JspServlet.service(JspServlet.java:241) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) org.apache.struts.action.RequestProcessor.doForward(RequestProcessor.java:1063) org.apache.struts.tiles.TilesRequestProcessor.doForward(TilesRequestProcessor.java:263) org.apache.struts.action.RequestProcessor.processForwardConfig(RequestProcessor.java:386) org.apache.struts.tiles.TilesRequestProcessor.processForwardConfig(TilesRequestProcessor.java:318) org.apache.struts.action.RequestProcessor.process(RequestProcessor.java:229) org.apache.struts.action.ActionServlet.process(ActionServlet.java:1194) org.apache.struts.action.ActionServlet.doPost(ActionServlet.java:432) javax.servlet.http.HttpServlet.service(HttpServlet.java:709) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) Thanks, _Martin

1 0

Kamailio Crashing when using Evapi with CGRateS
by Efelin Novak 18 Jun '19

18 Jun '19

Hi folks, I am doing some experiments with Kamailio and Cgrates. To be precise I am testing evapi module. When in testing environment I do not have any problem. After redirecting some production traffic to this kamailio, It crashes in at random times (few minutes to 20). I didn't find any pattern here. CGRateS doesn't write anything in logs. I'm on the last 4.4 Kamailio (compiled recently). CGRateS is on the last version. Can you please point to how to continue with debugging? Thanks Efelin Here is the problem in syslog: kamailio[9594]: : <core> [mem/f_malloc.c:599]: fm_free(): BUG: fm_free: bad pointer 0x97894818 (out of memory block!), called from tm: h_table.c: free_cell_helper(185) - abortingkamailio[9608]: CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 16 kamailio[9588]: ALERT: <core> [main.c:743]: handle_sigs(): child process 9594 exited by a signal 6 kamailio[9588]: ALERT: <core> [main.c:746]: handle_sigs(): core was not generated kamailio[9588]: INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD kamailio[9604]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received kamailio[9606]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received kamailio[9603]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received bt full #0 0xb77d1424 in __kernel_vsyscall () No symbol table info available. #1 0xb7673661 in raise () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7676a92 in abort () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 No symbol table info available. #3 0x082611a3 in fm_free (qmp=0x56779000, p=0x2d4e4f49, file=0x970932ee "tm: h_table.c", func=0x970935b7 "free_cell_helper", line=185, mname=0x97092d20 "tm") at mem/f_malloc.c:601 qm = 0x56779000 f = 0x56df046c __FUNCTION__ = "fm_free" #4 0x96fd0559 in free_cell_helper (dead_cell=0x56d935c4, silent=0, fname=0x970b90ca "timer.c", fline=648) at h_table.c:185 b = 0x56df0490 "CANCEL sip:00444123456@192.168.1.3:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKb6ae.5d03d7e94e5ec25f34930629f434a006.1\r\nMax-Forwards: 66\r\nFrom: \"0044454321\" <sip:0044454321@i"... i = 1 rpl = 0x9797e818 tt = 0xbfe44c68 foo = 0x970ccb14 cbs = 0x0 cbs_tmp = 0x56d12b68 __FUNCTION__ = "free_cell_helper" #5 0x9706884f in wait_handler (ti=31715179, wait_tl=0x56d9360c, data=0x56d935c4) at timer.c:648 p_cell = 0x56d935c4 ret = 0 #6 0x08241e20 in timer_list_expire (t=31715179, h=0x567a3b68, slow_l=0x567a5394, slow_mark=1772) at timer.c:873 tl = 0x56d9360c ret = 0 #7 0x08242211 in timer_handler () at timer.c:938 saved_ticks = 31715179 run_slow_timer = 0 i = 748 __FUNCTION__ = "timer_handler" #8 0x0824266b in timer_main () at timer.c:977 No locals. #9 0x080f0005 in main_loop () at main.c:1699 i = 3 pid = 0 si = 0x0 si_desc = "udp receiver child=2 sock=192.168.1.2:5060\000\270\301\310V\270\301\310V\000\000\000\000\000\000\000\020c\000\000\000\001\000\000\000\000\000\000\000\374Q\344\277HO\344\277\341`&\b\230\071\220\227\004\000\000\000٧5\b$\271\065\bn\000\000\000\324\247\065\b\n\000\000\000\363\320\372\060", '\000' <repeats 11 times> nrprocs = 3 woneinit = 1 __FUNCTION__ = "main_loop" #10 0x080f6b3a in main (argc=13, argv=0xbfe45374) at main.c:2644 cfg_stream = 0x8491008 c = -1 r = 0 tmp = 0xbfe4577c "" tmp_len = 1 port = 0 proto = 0 options = 0x8350e74 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:" ret = -1 seed = 1759582648 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbfe452a0 p = 0x805cc8c "[\201Ä4;" st = {st_dev = 14, __pad1 = 0, st_ino = 6161, st_mode = 16877, st_nlink = 2, st_uid = 109, st_gid = 111, st_rdev = 0, __pad2 = 0, st_size = 60, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1560160293, tv_nsec = 983219454}, st_mtim = {tv_sec = 1560165082, tv_nsec = 861178334}, st_ctim = {tv_sec = 1560165082, tv_nsec = 861178334}, __unused4 = 0, __unused5 = 0} __FUNCTION__ = "main"

2 2

Modify DMQ 200OK in on_reply route
by jenus＠cyberchaos.nl 18 Jun '19

18 Jun '19

Hi all, I'm trying to run a kamailio DMQ node inside a docker container. All works fine but i have one issue with DMQ. Since the kamailio process inside the container is behind a nat i have set the advertise address to the external ip, this works fine for the via headers but DMQ keeps sending the internal interface in the KDMQ notification_peer messages. The external ip is in there as well. Example: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.13;branch=z9hG4bK2dcb.c6326de2000000000000000000000000.0 To: <sip:notification_peer@10.10.10.94:5060>;tag=616b7acb681ed5e9cbc485b3faa09d62.0a3d From: <sip:notification_peer@10.10.10.13:5060>;tag=e8f29c06c4b85a188f6533f08d60feb6-ff29 CSeq: 10 KDMQ Call-ID: 17042656602ca532-23367(a)x.x.x.x Content-Type: text/plain Server: kamailio (5.1.4 (x86_64/linux)) Content-Length: 222 sip:10.10.10.13:5060;status=active sip:10.10.10.12:5060;status=active sip:10.10.10.92:5060;status=active sip:10.10.10.93:5060;status=active sip:10.10.10.94:5060;status=active <-- entry for container external ip sip:172.17.0.2:5060;status=disabled <-- entry for container internal ip I tried to strip this entry using the replace_body_atonce in the tm:local-request route. This works fine for the KDMQ request, but when i try to remove the same line from 200OK responses that are send out as a response to incomming KDMQ messages this seems to fail. I use the following code: # Working code (for outgoing KDMQ requests): event_route [tm:local-request] { if(is_method("KDMQ") && $rU == "notification_peer") replace_body_atonce("sip:172.17.0.*:5060;status=.*", ""); } # Failing code (for 200OK responses): reply_route { xlog("L_ALERT", "REPLY: Sending out reply, tU is $tU rs is $rs\n"); if ($rs == "200" && $tU == "notification_peer") { replace_body_atonce("sip:172.17.0.*:5060;status=.*", ""); } } I'm running kamailio 5.1.4. Is there a way to strip a line from the body of a local generated 200OK response? Or is there a way to avoid that DMQ inserts the internal container ip in the first place. Thanks, Jan Hazenberg

3 11

Rabbitmq module blocks sip request
by Denys Pozniak 18 Jun '19

18 Jun '19

Hello! I use rabbitmq module to get destination IP for the next call routing, but this module blocks sip request in case of RabbitMQ server is unreachable (timeout_sec/usec does not run out here). - Is there any solution to avoid such behavior and force function rabbitmq_publish_consume() to return an error code and continue script execution? - Is this connected with the nature of rabbitmq-c client? ("Currently librabbitmq offers no async API, but sync API, with a timeout.") -- BR, Denys Pozniak

1 0

Info: Kamailio.org - Infrastructure Maintenance - June 18, 2019
by Daniel-Constantin Mierla 18 Jun '19

18 Jun '19

The infrastructure hosting the main server for Kamailio project will go under scheduled maintenance during June 18, 2019 (tomorrow). The work is expected to start around 7:00am UTC and can last for a few hours. During the maintenance interval, the services related to Kamailio project may become unavailable, among them: * the web server (www.kamailio.org), including the wiki portal (www.kamailio.org/wiki) * the mailing lists and the email contact addresses * the GIT repository mirror git.kamailio.org Once the maintenance work is completed, there will be an update notification sent about it! If there are relevant events to be notified, we are going to publish on our twitter stream: * https://twitter.com/kamailio Also, you can join #kamailio channel on irc.freenode.net in case you want to report some incident related to our infrastructure that you discover while maintenance goes on. For bugs or other issues related to Kamailio code, use the tracker on github.com/kamailio/kamailio – that service is not affected being maintained by GitHub corporation. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

Kamailio 4.4.x CDR db accounting
by Laura 18 Jun '19

18 Jun '19

Dear guys, I'm looking to understand the "info" field inside the CDR written on acc table for Kamailio 4.4.x.. It seem that the field is never written with datas.. Is there a way to write some informations inside it ? Thanks for any reply

3 3

Forward calls between kamailio registrar nodes (dmq + dmq_usrloc)
by Ildar Mukhametzhanov 18 Jun '19

18 Jun '19

Hello! I am developing setup with several kamailio nodes as registrar cluster (dmq + dmq_usrloc). Location sync works, I see locations of both servers on every server. Is there any standard approach to redirect call between nodes except replacing of $ru? Executing of lookup("location") for user registered on another node return negative response. -- Sincerely, *Ildar Mukhametzhanov* Head of software development VoiceON Telecom Services GmbH im(a)voice-on.net <mailto:im@voice-on.net> t. +49 30 398 20 500-77

1 0

Authenticating xhttp request
by Olli Attila 18 Jun '19

18 Jun '19

Hello, I have this xhttp event_route on Kamailio that I am using to signal the proxy to reload dialplans and htable when necessary: event_route[xhttp:request] { if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Now instead of returning 403 forbidden for requests coming from other src_ip than proxy itsef, I would like to authenticate the http request via proxy database. How can this be done if possible? Cheers, Olli

3 7

dispatcher and TLS targets
by Karsten Horsmann 17 Jun '19

17 Jun '19

Hi all, i try to configure an SBC OS config [1] based kamailio 5.2.3 [2] with dispatcher and rtpengine. I used transport=tcp to see the plain traffic and then switched to TLS (with tls.cfg, valid certificate and stuff). After starting up, the Target is marked as "down". Due the encryption its hard to debug that. Any hints? Did i made an mistake in the configuration? TLS calls from the target to my kamailio proxy works. So its "half broken" :) at the moment. [1] https://github.com/voiceboys/sbcOS/blob/master/SbcOS/configs/voice/kamailio… [2] kamailio -v version: kamailio 5.2.3 (x86_64/linux) c36229 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: c36229 compiled on 11:28:11 May 22 2019 with gcc 4.8.5 -- %< --------------------- kamctl dispatcher dump "SET": { "ID": 1004, "TARGETS": [{ "DEST": { "URI": "sip:sip101.example.de;transport=tls", "FLAGS": "TP", "PRIORITY": 0, "ATTRS": { "BODY": "access=212.xx.xx.xx:5061;socket=tls:212.xx.xx.xx:5061;weight=100;ping_from=sip: mykamailio.example.de", "DUID": "", "MAXLOAD": 0, "WEIGHT": 100, "RWEIGHT": 0, "SOCKET": "tls:212.xx.xx.xx:5061" }, "LATENCY": { "AVG": 30000, "STD": 0, "EST": 30000, "MAX": 30000, "TIMEOUT": 1 } } }] } }, -- %< --------------------- kamctl dispatcher dump WARNING: <script>: Destination down: OPTIONS ru=sip101.example.de;transport=tls du=<null> -- %< --------------------- tls.cfg [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de [server:212.xx.xx.xx:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client: 212.xx.xx.xx:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de [client:default] verify_certificate = no require_certificate = no -- %< --------------------- tls.cfg Cheers Karsten -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

sr -user unsubscribe request
by Martin Lane 17 Jun '19

17 Jun '19

Please unsubscribe me from the sr-users list. V/R, Martin

1 0

Dispatcher's probing malfunction
by Duarte Rocha 17 Jun '19

17 Jun '19

Greetings, I have some questions about dispatcher's probing mechanics : 1 - In my SBC i have 26 gateways being probed. However, if i do a SIP capture no more than 16 gateways are probed each minute. Some gateways are always probed, others only once in a while. 2 - I have two SBC working in a pair with DB being replicated and with HA IPs. That being said i configure one NAP twice, each configuration with one HA socket. Is this configuration supported? I get very weird behaviours when i configure it like this. Best Regards, Duarte Rocha

1 0

restrict calls to only registered users
by Gaurav Bmotra 17 Jun '19

17 Jun '19

hi i m using kamailio 5.1 , with UBUNTU 18.1 i want to restrict call to only registered user ,,, plz help me thank you -- *Regards:* Gaurav Kumar

3 2

Querying in-memory location table (dmq_usrloc)
by Gholamreza Sabery 16 Jun '19

16 Jun '19

I am using dmq_usrloc, to replicate user locations across multiple Kamailio instances. Everything works fine. The problem occurs when a user on Kam1 wants to call a user on Kam2. How can I query the in-memory location table to see where the user is registered? PS: I use a Path header to specify the server that each user is registered on.

2 2

Softswitch
by Mary Honey 16 Jun '19

16 Jun '19

Hello, Is it possible to use Kamailio as a grade 5 Softswitch ? I need to handle 1000 calls/s

2 1

Stateless message forwarding with forward() add extra identical Via header.
by Ali Shirvani 15 Jun '19

15 Jun '19

Hi all, According to this SO question <https://stackoverflow.com/questions/32519551/kamailio-concurrent-calls-afte…> I tried to forward received message to multiple destinations with the following Lua script. But in the second UA I got duplicate Via header. Here is the Lua script: ```Lua KSR.forward_uri("sip:ali@192.168.1.50:7080;transport=UDP"); KSR.forward_uri("sip:tes@192.168.1.50:9080;transport=UDP"); ``` And here is the received Via headers is `tes` UA: ``` Via: SIP/2.0/UDP 192.168.1.21;branch=z9hG4bKbe8c.953ed42f0bcc438c6991121d3483010b.0 Via: SIP/2.0/UDP 192.168.1.21;branch=z9hG4bKbe8c.953ed42f0bcc438c6991121d3483010b.0 Via: SIP/2.0/UDP 10.189.219.4:5060 ;rport=5060;rport=5060;branch=z9hG4bK-4177-1-2 ``` Would you please help me to forward message to multiple destination without duplicate Via headers? Regards, Ali

1 0

403 Forbidden
by Pejic, Miroslav 14 Jun '19

14 Jun '19

Hi :) I'm trying to save a failed call in the kamailio_router.acc_cdrs database after sending the 403 Forbidden by router, using the module acc, if is it possible? https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html Something like this: route[IS_CALLER_BLACKLISTED] { python_exec("check_caller_blacklisted"); if ($var(callerBlacklisted) == 1){ t_send_reply("403", "Forbidden"); IT’S POSSIBLE HERE SAVE A CALL ON KAMAILIO_ROUTER.ACC_CDRS? exit; } Thanks in advance! Miro

4 6

Re: [SR-Users] 403 Forbidden
by Federico Cabiddu 14 Jun '19

14 Jun '19

I understand, to get the cdr you should have created the dialog before replying. Cheers, Federico On Fri, 14 Jun 2019, 17:05 Pejic, Miroslav, <miroslav.pejic(a)cloud.timenet.it> wrote: > [image: image.png] > > The call doesn't even begin, I immediately block it. :) > > Il giorno ven 14 giu 2019 alle ore 16:59 Pejic, Miroslav < > miroslav.pejic(a)cloud.timenet.it> ha scritto: > >> Hi Federico, >> >> i tryied also with `modparam("acc", "cdr_on_failed", 1)` >> but in my case the call is not saved at all on `acc_cdrs`, >> with the second method i mentioned works :) >> Thanks >> Miro >> >> Il giorno ven 14 giu 2019 alle ore 16:14 Federico Cabiddu < >> federico.cabiddu(a)gmail.com> ha scritto: >> >>> Hi, >>> dialog CDR for failed calls can be generated enabling >>> >>> http://www.kamailio.org/docs/modules/devel/modules/acc.html#acc.p.cdr_on_fa… >>> >>> It has been some time since I last tested it, but it still should work :) >>> >>> Cheers, >>> >>> Federico >>> >>> On Fri, Jun 14, 2019 at 3:25 PM Daniel-Constantin Mierla < >>> miconda(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> it should be ok also with 2) if it gives what you want. Kamailio is >>>> just writing to acc_cdrs, if you have extra columns, it should not be >>>> affected. >>>> >>>> Cheers, >>>> Daniel >>>> On 14.06.19 14:16, Pejic, Miroslav wrote: >>>> >>>> Hi Daniel :) >>>> >>>> i tryied 2 methods to save it >>>> >>>> 1) i created a new table on kamailio_router called `acc` >>>> >>>> from `utils/kamctl/mysql/acc-create.sql` >>>> >>>> ``` >>>> CREATE TABLE `acc` ( >>>> `id` INT(10) UNSIGNED AUTO_INCREMENT PRIMARY KEY NOT NULL, >>>> `method` VARCHAR(16) DEFAULT '' NOT NULL, >>>> `from_tag` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `to_tag` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `callid` VARCHAR(255) DEFAULT '' NOT NULL, >>>> `sip_code` VARCHAR(3) DEFAULT '' NOT NULL, >>>> `sip_reason` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `time` DATETIME NOT NULL >>>> ); >>>> CREATE INDEX callid_idx ON acc (`callid`); >>>> ``` >>>> then >>>> >>>> ``` >>>> route[IS_CALLER_BLACKLISTED] { >>>> >>>> python_exec("check_caller_blacklisted"); >>>> >>>> if ($var(callerBlacklisted) == 1){ >>>> >>>> t_send_reply("403", "Forbidden"); >>>> >>>> acc_db_request("403 Forbidden", "acc"); >>>> >>>> exit; >>>> >>>> } >>>> >>>> ``` >>>> >>>> in `acc` i have now calls that were not answered with 200ok :) >>>> >>>> >>>> 2) add new fields on exsisting acc_cdrs table used to store cdrs >>>> >>>> ``` >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `from_tag` varchar(128) DEFAULT NULL >>>> AFTER `dst_domain`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `to_tag` VARCHAR(128) DEFAULT NULL >>>> AFTER `from_tag`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `sip_code` VARCHAR(3) DEFAULT NULL >>>> AFTER `to_tag`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `sip_reason` varchar(128) DEFAULT >>>> NULL AFTER `sip_code`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `time` datetime DEFAULT NULL AFTER >>>> `sip_reason`; >>>> ``` >>>> >>>> ``` >>>> route[IS_CALLER_BLACKLISTED] { >>>> >>>> python_exec("check_caller_blacklisted"); >>>> >>>> if ($var(callerBlacklisted) == 1){ >>>> >>>> t_send_reply("403", "Forbidden"); >>>> >>>> acc_db_request("403 Forbidden", "acc_cdrs"); >>>> >>>> exit; >>>> >>>> } >>>> >>>> ``` >>>> >>>> In this way i have both stored calls that were not answered with 200ok >>>> and that were answered with 200ok. >>>> The idea is to have all in one table `acc_cdrs`. >>>> >>>> Can be ok also the second solution? >>>> >>>> Thank you very much! >>>> Miro >>>> >>>> Il giorno ven 14 giu 2019 alle ore 09:06 Daniel-Constantin Mierla < >>>> miconda(a)gmail.com> ha scritto: >>>> >>>>> Hello, >>>>> >>>>> I don't think that you can get a dialog-based CDR for calls that were >>>>> not answered with 200ok. But you can get a record in acc table, by storing >>>>> failed transaction record (see the mod params for acc and failed >>>>> transaction flag or use acc_db_request()). >>>>> >>>>> Cheers, >>>>> Daniel >>>>> On 13.06.19 22:45, Pejic, Miroslav wrote: >>>>> >>>>> Hi :) >>>>> >>>>> >>>>> I'm trying to save a failed call in the kamailio_router.acc_cdrs >>>>> database >>>>> >>>>> after sending the 403 Forbidden by router, using the module acc, if is >>>>> it possible? >>>>> >>>>> >>>>> https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html >>>>> >>>>> >>>>> Something like this: >>>>> >>>>> >>>>> route[IS_CALLER_BLACKLISTED] { >>>>> >>>>> >>>>> python_exec("check_caller_blacklisted"); >>>>> >>>>> >>>>> if ($var(callerBlacklisted) == 1){ >>>>> >>>>> >>>>> t_send_reply("403", "Forbidden"); >>>>> >>>>> >>>>> IT’S POSSIBLE HERE SAVE A CALL ON KAMAILIO_ROUTER.ACC_CDRS? >>>>> >>>>> >>>>> >>>>> exit; >>>>> >>>>> } >>>>> >>>>> >>>>> Thanks in advance! >>>>> >>>>> Miro >>>>> >>>>> _______________________________________________ >>>>> 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 >>>>> >>>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>

1 0

Getting which timer has expired
by Igor Olhovskiy 14 Jun '19

14 Jun '19

Hi! Is there any way to get which timer on transaction actually expired? Scenario is following: INVITE - 180 - 200 OK - (restransmissions 250 msec starts) - pause 15 sec - ACK. In INVITE i'm setting t_on_reply("MY_REPLY") But in around 10 sec after first 200 OK (and 12 sec after INVITE) I'm getting in logs DEBUG: tm [h_table.c:133]: free_cell_helper(): freeing transaction 0x7fb632d77090 from timer.c:689 and on next 200 OK I'm never getting into onreply_route[MY_REPLY] again. Nowhere in the code or config I have explicit set any of timers with 10 or 12 seconds value. Most of them are default. So, can I somehow get which timer expired exactly? PS: If any, pause in 15 sec for a reason to emulate this issue. Regards, Igor

2 2

Event Route not found when loading XHTTP module
by Dhruva TS 14 Jun '19

14 Jun '19

Hello, I am new to Kamailio and I am using Kamailio 5.2. When trying to the load the XHTTP module, event route not found error is thrown and XHTTP is module is not getting loaded. I am trying to use JSONRPC module through HTTP transport for RPC calls. Using XMLRPC, things work fine. DEBUG: <core> [core/sr_module.c:896]: init_mod(): xhttp ERROR: xhttp [xhttp_mod.c:136]: mod_init(): failed to find event_route[xhttp:request] ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module xhttp (/usr/lib/kamailio/modules/xhttp.so) ERROR: error while initializing modules I have added below to Kamailio.cfg file: loadmodule "xhttp.so" event_route[xhttp:request] { if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Any help in this regard. Thanks, Dhruva

2 1

S-CSCF replies “Forbidden - Domain not served”
by Bao, Yin 13 Jun '19

13 Jun '19

We have configured Kamailio IMS (P, I, S-CSCF) to use a single realm with the same NETWORKNAME and trying to make a call between bob and alice. We are getting S-CSCF replies “Forbidden – Domain not served” in two scenarios: 1. When bob calls alice, the INVITE went from P->S->I->S and the destination S-CSCF responded this error 2. During SIP registration in responding to SUBSCRIBE (endpoint still can register even with this reply, so we proceeded with making call) In quick search we found that in scscf/kamailio.cfg route{ } implementation there is a case when this error is generated: # Terminating if (uri == myself || uri =~ "tel:.*") { if (!term_impu_registered("location")) { xlog("L_DBG", "We need to do an UNREG server SAR assignemnt"); assign_server_unreg("UNREG_SAR_REPLY", "location", "term"); exit; } } else { sl_send_reply("403","Forbidden - Domain not served"); exit(); } route(term); break; What is this error designed for? Also in general are we supposed to rework the routing code in Kamailio.cfg (P, S, I-CSCF)? Thanks for any suggestions in advance. Yin

1 0

Problem with Kamailio 5.2.x Repository
by Joshua Riffle 13 Jun '19

13 Jun '19

Hi, Heads up... I suspect the Kamailio 5.2.x RPM files for CentOS 7 were recently updated June 10th and are now no longer accepted by a standard yum install... *Repository file in /etc/yum.repos.d/:* [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl= http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… gpgcheck=1 gpgkey= http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… enabled=1 *Operating System Version:* CentOS Linux release 7.6.1810 (Core) (Fresh image -- varying cloud providers) *Command to install:* yum install kamailio *Resulting error message:* kamailio-5.2.3-4.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.3-4.el7.centos.x86_64: [Errno 256] No more mirrors to try. *Troubleshooting:* I verified the IP address for the opensuse URL with three different Internet providers and tested the installation on servers in different cloud providers geolocated in both Canada and United States and received the exact same results in all cases. -- Joshua Riffle *DevOps Lead & Duke of the Dial Tone* We Are Skyetel and You Are Awesome. p: (360) 843-9903 o: (360) 986-5200 s: http://www.skyetel.com e: josh(a)skyetel.com

2 2

CRITICAL ERROR in data_lamp.c!
by Mojtaba 13 Jun '19

13 Jun '19

Hello, What does this CRITICAL error mean? Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: <core> [core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out of memory block!) called from core: core/data_lump.c: free_lump(466) - aborting I got it when i make new call during debugging new module in Kamailio? In module we have to change some message's body with data_lump. I asked relaed issue (strange behaviour while developing new module) in Kamailio developer forum <sr-dev(a)lists.kamailio.org> With Best Regards. -- --Mojtaba Esfandiari.S

2 10

dialog:end event route
by Igor Olhovskiy 13 Jun '19

13 Jun '19

Hi! Just to confirm, is event route [dialog:end] not called on fr_inv_timer expiry? Cause it's called on call busy (486) and reject (whatever 6xx). So, what's the idea of not calling it on timeout? Or I'm missing something? Kamailio 5.2 Thanks! Regards, Igor.

2 8

Best practises for websocket authentication
by Grant Bagdasarian 12 Jun '19

12 Jun '19

Hello, I was wondering if there are any best practices for authenticating websocket connection establishment inside the xhttp event route? Thanks and regards, Grant

3 3

TCP reset behavior
by R, Rajkumar (Raj) 12 Jun '19

12 Jun '19

Hi All, In our setup running kamailio (alpine image) in stateless mode and have following TCP config set. After client registers to the backend see two TCP connections created - one towards the client and the other towards the backendserver. If there's a abrupt network disconnect either at the client side or at the backend side, do not see TCP reset happening from kamailio as per the config. Per below config expected after 120seconds( = [tcp_keepidle] + ( [tcp_keepintvl * tcp_keepcnt] ) instead it keeps resending TCP keepalive packets at varying intervals for varying duration length(like 6mins, 14mins..) before resetting the TCP connections. Also want to mention that we are not using TCPOPS module here. I am trying to achieve consistent duration for TCP reset in kamailio. Could you please point me if I have missed anything here? Setup: Client ---- Kamailio (stateless mode) ---- Backend server TCP Config: tcp_connection_lifetime = 3600 tcp_accept_aliases = no tcp_async = yes tcp_connect_timeout = 10 tcp_crlf_ping = yes tcp_keepalive = yes tcp_keepcnt = 3 tcp_keepidle = 30 tcp_keepintvl = 30 # Total 2 mins - 30+(30*3)=120 sec tcp_linger2 = 30 tcp_rd_buf_size = 65536 tcp_send_timeout = 10 thanks, raj

2 9

Setting the probing state failed
by Daniel Greenwald 12 Jun '19

12 Jun '19

ERROR: dispatcher [dispatch.c:2635]: ds_options_callback() Setting the probing state failed (sip:10.0.9.133:5064, group 0) I get this message periodically from 3 kamailio dispatchers simultaneously for the same destination host. Any idea what this error means? Dispatcher settings below: modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "sock_avp", "$avp(dssocket)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "ds_hash_size", 3) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:pinger@MYPRIVATEIP :INTERNALPORT") #modparam("dispatcher", "ds_ping_method", "INFO") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")

2 1

How reply INVITE request with 200 OK that contains SDP?
by Ali Shirvani 11 Jun '19

11 Jun '19

Hi all, I tried to implement simple announcer by using play_media function in latest rtpengine module. I implemented simple Lua script and I tried to test the implementation with simple SIPp scenario. Here is the Lua function to reply the received INVITE request. ```Lua function ksr_route_announcer() KSR.info("======= Announcer =====\n"); local u = KSR.pv.gete("$rU"); KSR.info("username is: " .. u .. "\n"); if u == "ann" then KSR.info("announcer has been called.\n"); if KSR.textops.has_body_type("application/sdp") then KSR.info("request contains SDP.\n"); KSR.tm.t_newtran(); KSR.rtpengine.rtpengine_manage0(); KSR.tm.t_reply(200, "OK"); KSR.x.exit(); end end end ``` According to kamailio logs rtpengine modified the INVITE SDP, but I want to add SDP to 200 OK response. How should I manage rtpengine to append SDP answer to the 200 OK response? The attached is the SIPp log, as you can see that last 200 OK doesn't contain SDP. Regards, Ali

2 1

Re: [SR-Users] [EXT] Re: Kamailio 5.2.2 ICSCF changing sip to tel
by Woscek, Martin W. 11 Jun '19

11 Jun '19

Daniel, Actually, the HSS is not returning the uri with the +. There are no users int the HSS database with a public ID with a + so the HSS returns the USER_UNKNOWN. Looking at the PCAP, the I-CSCF sends a diameter msg to the HSS asking for the verification the public ID of: Public-Identity: sip:+alice@ims522.irisims.org The diameter msg from HSS is returning : AVP: Experimental-Result-Code(298) l=12 f=-M- val=DIAMETER_ERROR_USER_UNKNOWN (5001) Why do we want the I-CSCF to add a + to all URI's don't start with a +? The I-CSCF is the module that is adding the +, assuming that we should be able to included Kamailio.cfg for respective modules, is there a directive or variable that I need to set? Or do I simply comment out that code? Thanks, Martin Woscek The MITRE Corp. mwoscek(a)mitre.org<mailto:mwoscek@mitre.org> Phone: 703-983-2650 FAX: 703-983-7142 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Tuesday, June 11, 2019 6:22 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Woscek, Martin W. <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio 5.2.2 ICSCF changing sip to tel Hello, that prefix operation is done only if the r-uri username does not have +. So if you HSS is returning the r-uri with +, then all is ok, no other + is added. On the other hand, I am not sure what is the error or the issue you try to sort out. The error message is because of the xlog(...) in the config. Cheers, Daniel On 10.06.19 16:17, Woscek, Martin W. wrote: Hello all, The HSS is responding with 604 errors when the I-CSCF checks the public identity of the caller. PCAP attached. It looks like our I-CSCF is modifying the sip uri's to tel uri's (e.164-like), referenced attached PCAP: in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.irisims.org<mailto:alice@ims522.irisims.org> syslog output: Jun 6 16:50:06 imskamailio /usr/local/sbin/kamailio[14066]: ERROR: <script>: $ru => sip:+alice@irisims.org Looking at the default (out of the box) Kamailio.cfg for I-CSCF the following script snipet is where the modification to our sip URI is happening: # ------------------------- request routing logic ------------------- # main routing logic route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } Is there a I-CSCF setting directive we missed? Or should we simply comment the above? Any advice is appreciated. Thanks, Martin Woscek _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>

2 1

Relay NAP Probing
by Duarte Rocha 11 Jun '19

11 Jun '19

Greetings, I have a Kamailio as Sip Proxy between my clients and my Media gateway. I have NAPs (One per client) for clients and NAPs for the MediaGateway (One for each client). The MediaGateway probes Kamailio in order to see if it's alive and Kamailio probes clients. However, i would like for the MediaGateway to be aware of the client status. I thought of a few solutions : - When Kamailio probes a client and it fails, both NAPs (Client Side and MG side) could be set as inactive. - Automatically relay the MG probe to the client - Some scripting that reads the result of dispatcher.list and if it finds an inactive NAP, set the MG NAP as inactive too. Is there a better/simpler way to do this? I'd rather have everything on Kamailio configuration files instead of doing exatra scripting. Any opinions ? How can i achieve this? Best Regards, Duarte Rocha

2 1

Kamailio 5.2.2 ICSCF changing sip to tel
by Woscek, Martin W. 11 Jun '19

11 Jun '19

Hello all, The HSS is responding with 604 errors when the I-CSCF checks the public identity of the caller. PCAP attached. It looks like our I-CSCF is modifying the sip uri's to tel uri's (e.164-like), referenced attached PCAP: in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.irisims.org syslog output: Jun 6 16:50:06 imskamailio /usr/local/sbin/kamailio[14066]: ERROR: <script>: $ru => sip:+alice@irisims.org Looking at the default (out of the box) Kamailio.cfg for I-CSCF the following script snipet is where the modification to our sip URI is happening: # ------------------------- request routing logic ------------------- # main routing logic route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } Is there a I-CSCF setting directive we missed? Or should we simply comment the above? Any advice is appreciated. Thanks, Martin Woscek

2 1

Operations after relay
by Nicolas Breuer 11 Jun '19

11 Jun '19

Hello, I want to do some sql operations after a 200 reply, after relaying the sip packet Is it possible or sending the sip packet is the last operation permitted ?

3 3

Connect SIP Trunk to PBX
by Chad Kenealy 10 Jun '19

10 Jun '19

Hello, I hope you can help, We are trying to connect our SIP trunk to our new PBX system, however our SIP trunk provider requires a static IP to add to the endpoint. Our new PBX provider uses a range of IPs that are connected to a domain and the IPs change all the time. Is there a way of using Kamailio that we could get our Sip Trunk provider and PBX system to work together? Regards

2 1

Question about setbflag and isbflagset usage
by Володимир Іванець 07 Jun '19

07 Jun '19

Hello everyone! I'm not sure if I correctly understood the description of functions *isbflagset* and *isbflagset* so once again I'm asking for help :) Here is what I do: 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of available contacts and then few checks in branch_route. I'm adding *setbflag("17", $T_branch_idx);* there. 2. Later in route[NATMANAGE] I added *if (isbflagset("17", $T_branch_idx)) {* and additional logic. *isbflagset* results in *true* (for same INVITE). 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK packet that gets in route[NATMANAGE]. *isbflagset* results in *false*. So my question is: shouldn't *isbflagset* be *true* for ACK too since it's the same branch or I interpreted these fenctions incorrectly? Thanks a lot!

3 6

Re: [SR-Users] [EXT] Re: ICSCF mysql error
by Bao, Yin 07 Jun '19

07 Jun '19

Yes I think you are right it’s the mismatch of uri. However we don’t know which configuration resulted that mismatch. Here are a bit more details: ICSCF prepared this query: select * from impu where identity=? And it provided sip:+alice@irisims.org as the parameter for the query However the HSS impu table as well as the SIP INVITE only contains: sip:alice@irisims.org I am suspecting the additional + may be causing this mismatch. But where is the + coming from and why ICSCF add this in front of original INVITE line uri? Thanks! Yin From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of YASIN CANER <caner_yaso(a)hotmail.com> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Friday, June 7, 2019 at 2:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] ICSCF mysql error Hello , It is a mismatch inconming uri and your uri in hss. Yasin Caner Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> şunları yazdı (7 Haz 2019 09:28): Hello, not an expert on this particular module, but this is the internal error path from the module that probably caused this error reply: - case RC_IMS_DIAMETER_ERROR_USER_UNKNOWN, will send message with: - #define MSG_604_USER_UNKNOWN "Does not exist anywhere - HSS User Unknown" The status comes probably from the CDP module: diameter_epc_code_result.h 57: RC_EPC_DIAMETER_ERROR_USER_UNKNOWN = 5001, Maybe this helps. Henning Am 03.06.19 um 06:43 schrieb Trinh, Jenny: Hi, I’m getting the error 604 when ICSCF did a query to mysql and failed with an INVITE. Please see my attached pcap. Would appreciate any feedbacks on what might the issue. We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: SIP/2.0/UDP 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: <sip:bob@ims522.irisims.org><mailto:sip:bob@ims522.irisims.org>;tag=594048364#015#012To: <sip:alice@ims522.irisims.org><mailto:sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 req_src_ip=10.0.20.1 req_src_port=6060 in_uri=sip:alice@ims522.irisims.org<mailto:in_uri=sip:alice@ims522.irisims.org> out_uri=sip:+alice@ims522.ir<mailto:out_uri=sip:+alice@ims522.ir> Thank you _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] Using DNS in Dispatcher for SRV Clarification [SOLVED]
by JR Richardson 07 Jun '19

07 Jun '19

Hi Federico, "use_dns_cache=on" this is on be default. "use_dns_failover=on" this is off by default, changing to "on" did the trick, auto failover to next SRV server is working quickly, as expected now. Thanks for the pointer. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope > > Hi, > if you want failover support, you need to have a couple of core dns > parameters enabled: > https://www.kamailio.org/wiki/cookbooks/5.2.x/core#use_dns_cache > > and > > https://www.kamailio.org/wiki/cookbooks/5.2.x/core#use_dns_failover > > The latter defaults to off. > > Cheers, > > Federico > > On Thu, Jun 6, 2019 at 10:10 PM JR Richardson <jmr.richardson(a)gmail.com> > wrote: > > > Hey All, > > > > I'm in the lab testing kamailio 5.2 with DNS SRV records. The scenario > > is simple, using dispatcher with hard coded IP URIs > > [sip:10.10.10.10:5060] and also a DNS name URI [sip:sip.domain.com]. > > There are 2 SRV servers. > > > > The DNS SRV records are setup and resolving OK, during startup, > > kamailio list the DNS name in the dispatcher list. > > > > During a call scenario, when the dispatcher selects the DNS name URI > > for the next destination, the TM module takes over, resolves the > > _sip._udp.sip.domain.com for both SRV records OK and sends the call to > > the first SRV server in the list. But if I take the first SRV server > > offline, make another call, the TM module does not route to the second > > SRV server until the DNS record timeout is reached, in my lab was set > > for 300 seconds. I decreased the DNS SRV record TTL down to 5 seconds > > with same results above, but 5 seconds lapse when TM routes call to > > next SRV server. > > > > I think this is working as expected but not sure if this is by design > > so my question is, should the TM module route to another SRV server in > > the list in the event of a no response timeout or send the transaction > > back to the dispatcher to select another target? > > > > Should I play around with the TM Module parameter > > "max_noninv_lifetime" to get quicker results or it is better to have > > DNS SRV record TTL set really low? > > > > Thanks. > > > > JR

1 0

reading php script result with exec_avp
by Gertjan Wolzak 07 Jun '19

07 Jun '19

Hello Kamailions, I have the following challenge. I want to create a time dependent callerid, based on the time of a outgoing call a callerid will be set. I have a php script, that provides me an id number which relates to an cli. When I run that php script from the console, the id is shown in the console. When I run the script within the kamailio configuration, exec_avp("../script.php", "$avp(s:cliId)"); ,no output commande is saved in the avp variable. I think I am not presenting the result of the php script correctly. I have tried with an echo $variable; and with and fwrite(STDOUT, $variable); but no luck. I'm not a PHP coder so I am not surprised of the missing luck, but maybe somebody has a golden tip for me? Rgds, Gertjan

2 7

Re: [SR-Users] ICSCF mysql error
by Henning Westerholt 07 Jun '19

07 Jun '19

Hello, not an expert on this particular module, but this is the internal error path from the module that probably caused this error reply: - case RC_IMS_DIAMETER_ERROR_USER_UNKNOWN, will send message with: - #define MSG_604_USER_UNKNOWN "Does not exist anywhere - HSS User Unknown" The status comes probably from the CDP module: diameter_epc_code_result.h 57: RC_EPC_DIAMETER_ERROR_USER_UNKNOWN = 5001, Maybe this helps. Henning Am 03.06.19 um 06:43 schrieb Trinh, Jenny: Hi, I’m getting the error 604 when ICSCF did a query to mysql and failed with an INVITE. Please see my attached pcap. Would appreciate any feedbacks on what might the issue. We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: SIP/2.0/UDP 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: <sip:bob@ims522.irisims.org><mailto:sip:bob@ims522.irisims.org>;tag=594048364#015#012To: <sip:alice@ims522.irisims.org><mailto:sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 req_src_ip=10.0.20.1 req_src_port=6060 in_uri=sip:alice@ims522.irisims.org<mailto:in_uri=sip:alice@ims522.irisims.org> out_uri=sip:+alice@ims522.ir<mailto:out_uri=sip:+alice@ims522.ir> Thank you _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 1

Using DNS in Dispatcher for SRV Clarification
by JR Richardson 07 Jun '19

07 Jun '19

Hey All, I'm in the lab testing kamailio 5.2 with DNS SRV records. The scenario is simple, using dispatcher with hard coded IP URIs [sip:10.10.10.10:5060] and also a DNS name URI [sip:sip.domain.com]. There are 2 SRV servers. The DNS SRV records are setup and resolving OK, during startup, kamailio list the DNS name in the dispatcher list. During a call scenario, when the dispatcher selects the DNS name URI for the next destination, the TM module takes over, resolves the _sip._udp.sip.domain.com for both SRV records OK and sends the call to the first SRV server in the list. But if I take the first SRV server offline, make another call, the TM module does not route to the second SRV server until the DNS record timeout is reached, in my lab was set for 300 seconds. I decreased the DNS SRV record TTL down to 5 seconds with same results above, but 5 seconds lapse when TM routes call to next SRV server. I think this is working as expected but not sure if this is by design so my question is, should the TM module route to another SRV server in the list in the event of a no response timeout or send the transaction back to the dispatcher to select another target? Should I play around with the TM Module parameter "max_noninv_lifetime" to get quicker results or it is better to have DNS SRV record TTL set really low? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Re: [SR-Users] ICSCF mysql error
by Daniel-Constantin Mierla 06 Jun '19

06 Jun '19

Hello, check you kamailio configuration file, the 604 reply is likely to be generated there -- although I haven't written any of the ims modules, typically the reply generated by kamailio are from configuration file, not hardcoded in the source. Cheers, Daniel On 03.06.19 06:43, Trinh, Jenny wrote: > > Hi, > > I’m getting the error 604 when ICSCF did a query to mysql and failed > with an INVITE. Please see my attached pcap. Would appreciate any > feedbacks on what might the issue. > > > > We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 > > > > May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: > <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): > copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not > exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP > 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: > SIP/2.0/UDP > 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: > <sip:bob@ims522.irisims.org>;tag=594048364#015#012To: > <sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: > 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 > INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: > 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 > req_src_ip=10.0.20.1 req_src_port=6060 > in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.ir > > > > Thank you > > > _______________________________________________ > 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

1 0

Use dmq module with private and public IP addresses
by Markus Bönke 05 Jun '19

05 Jun '19

Hello NG, I have a question regarding the dmq / dmq_userloc module. In my test setup have 2 Kamailio servers in 2 different data centers. The data centers are interconnected by a crosslink. Each Kamailio server has configured an internal (private) and an external (public) IP address. The internal IPs are routed to use the crosslink. The external IPs are routed through the public internet by different ISPs. The requirement is that the replication should primarily be routed via the crosslink, only if the crosslink is down it should be routed via public internet. As I understand the „server_address“ parameter can have only one IP address. If I configure the public IPs on both sides and one ISP connection gets disconnected, packets will NOT be routed automatically via the crosslink. And the other way around, if I configure the private IPs on both sides and the crosslink gets disconnected, packets will NOT be routed automatically via public internet. Is there any way to setup and use dmq for such a scenario ? Thanks and regards Markus

2 3

IMS - change the transport layer protocol for Diameter based interfaces (especially Rx)
by "björn bauer" 05 Jun '19

05 Jun '19

3 2

RTPEngine and dialog moving.
by Igor Olhovskiy 04 Jun '19

04 Jun '19

Hi! Wanted to clarify some info about rtpengine working within HA-environment. Idea is we have multiple rtpengines under same setid. So, when I'm calling rtpengine_offer() I really don't care which is selected. But than might happen situation, where this call(dialog) is moved to other Kamailio instance (via dlg_db_load_callid()) and from there called rtpengine_delete(). So, my question is, how I can be sure, that delete request will go to same rtpengine instance that was selected in rtpengine_offer() on different server. And where this info is stored, within dialog or within TM module (which is not replicated in any way) Regards, Igor

3 4

Info: app_lua_sr module - old Lua sr library
by Daniel-Constantin Mierla 04 Jun '19

04 Jun '19

Hello, several days ago I split the app_lua module, relocating the code related to old Lua sr library to app_lua_sr module, keeping in this way app_lua smaller and focused to the KEMI code. Over the time, app_lua_sr should be deprecated and removed, as we are sure the functions in Lua sr have an equivalent in Lua KSR. Therefore, if you are using in Lua scripts functions from 'sr' library, consider migrating to KSR as soon as possible. The intermediary solution for a quick upgrade to master version (to be release as v5.3.x) is to load app_lua_sr and update some modparams, like: loadmodule "app_lua.so" loadmodule "app_lua_sr.so" modparam("app_lua_sr", "register", "sl") If you find a function from Lua sr not available in Lua KSR, then report it via mailing lists or bug tracker. An article about this topic was published at: * https://www.kamailio.org/w/2019/05/new-module-app_lua_sr/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Multiple definitions with OR or AND in "ifdef" in kamailio script
by Maxim M 04 Jun '19

04 Jun '19

Is it possible to do something like this in kamailio script?: !!ifdef DEF1 || DEF2 .... !!endif

2 1

empty action lists in route block
by Maryam Baghdadi 04 Jun '19

04 Jun '19

Hello everyone! I tried to use from Lua scripting language in kamailio configuration and rewrite all routes in Lua script. and I export many functions in ims_auth in KEMI and recompile ims_auth module. I have a problem with www_ims_challenge function calling in Lua script. when I define and call a function (REG_MAR_ROUTE) in www_ims_challenge as a route black, I get this error " empty action lists in route block" . why this happens and what's the solution ? thanks in advance

3 2

Removal of Dispatcher
by Colin, Andrew 04 Jun '19

04 Jun '19

I am using the API to remove dispatchers with a simple curl command, is there a way i can make kamailio check that there are sufficient dispatchers available before removing them when the command is run? Example: i am removing asterisk1 out of kamailio dispatcher list, but kamailio needs to ensure that there is atleast 2 or 3 other asterisk servers active in the list before it allows it to be removed. any ideas how i can achieve this?

2 3

AAR change subscription ID
by eyas barhouk 03 Jun '19

03 Jun '19

dear community, when i'm trying to deploy kamailio IMS with PCRF i get ims-qos AAR error due to Subscription id. in ims_qos module i found that i can change the subscription_id type and value as following: * subscription_id to hard code subscription ID for AAR. Used for some broken PCRFs. Leave blank to use default * subscription_id_type to hard code subscription ID type for AAR. Only applicable if subscription_id is set. Set to -1 to use default. This is as per RFC 4006: END_USER_E164 0, END_USER_IMSI 1, END_USER_SIP_URI 2, END_USER_NAI 3, END_USER_PRIVATE 4 but when i changed it there was no change on AAR Subscription-id AVP. any advise about how to set or change the subscription id dears ?? B.R Eyas

1 0

Releasing a new version for Siremis
by Daniel-Constantin Mierla 03 Jun '19

03 Jun '19

Hello, after quite a bit of time, we are considering to release a new version for Siremis - the web management interface for Kamailio. A lot of effort has been pushed to address the issues with PHPv7.x, which deprecated a lot of PHPv5.x API as well as made more strict rules in regards to classes and inheritances. At this stage it should be in good stage, but it would help if others can give it a ride to Siremis master branch: * https://github.com/asipto/siremis Installation guidelines: * https://kb.asipto.com/siremis:install51x:main Testing with both PHP v5 and v7 is appreciated in order to be sure things were not broken for old systems. Any issue found should be reported on github project issue tracker. If all goes fine, in soon we will make the release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

invalid payload location
by Kelvin Chua 31 May '19

31 May '19

is there a plan to create an sdp sanity functionality? ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location opensips has something similar sipmsg_validate() *'s'* - checks the integrity of the SDP body, if it exists Kelvin Chua

2 2

Table creation error when enabling ims_usrloc_scscf module.
by Ali Shirvani 30 May '19

30 May '19

Hi all, I installed kamailio v5.2.3 from source and IMS modules included. Then I tried to create required database and tables, I removed "dialog" from STANDARD_MODULES and added "ims_dialog ims_charging ims_usrloc_scscf ims_usrloc_pcscf presence" to the EXTRA_MODULES in kamctlrc file. When I issue `<kamailio_home>/sbin/kamctl create` I got following error: ERROR 1062 (23000) at line 39: Duplicate entry 'subscriber' for key 'table_name_idx' ERROR: Creating extra tables failed at ims_usrloc_scscf! Should I exclude an other module from STANDARD_MODUELS? Regards, Ali

2 1

turn on DEBUG level debugging for ICSCF module
by Trinh, Jenny 30 May '19

30 May '19

Hi, How do I turn on DEBUG level debugging for ICSCF module? Thanks Jenny

2 1

app_java.so: undefined symbol: JNI_CreateJavaVM please help
by Renato Gallo 30 May '19

30 May '19

gcj is not a requirement since 2018 [ https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html | https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html ] all details are in the bug ticket at the link: [ https://github.com/kamailio/kamailio/issues/1969 | https://github.com/kamailio/kamailio/issues/1969 ] please help Renato Gallo System Engineer sede legale e operativa: Via Privata Cefalonia, 14 - 20156 - Milano (MI) Tel. +39 02 - 87049490 Fax +39 02 - 48677349 Mobile. +39 342 - 6350524 Wi | FreeNumbers: https://freenumbers.way-interactive.com Wi | SMS: https://sms.way-interactive.com Wi | Voip: https://voip.way-interactive.com Asterweb: http://www.asterweb.org Le informazioni contenute in questo messaggio e negli eventuali allegati sono riservate e per uso esclusivo del destinatario. Persone diverse dallo stesso non possono copiare o distribuire il messaggio a terzi. Chiunque riceva questo messaggio per errore è pregato di distruggerlo e di informare immediatamente [ mailto:info@sigmaware.it | info@ ] asterweb.org

1 0

Fwd: Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 30 May '19

30 May '19

Hi Daniel, See pastebin url below for the backtrace. https://pastebin.com/tTUUk4YH Hopefully I did it the right way. Please let me know if you need anything else. FYI, Kamailio is running inside of a docker container. version: kamailio 5.2.1 (x86_64/linux) 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: unknown compiled with gcc 6.3.0 I can easily replicate this, but to be sure I've saved the core file locally on my machine. Regards, Grant On Thu, May 2, 2019 at 10:05 AM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > can you get a core file for this case? > > Set `ulimit -c unlimited` before starting Kamailio as root and run again. > > Once you have the core file, get the backtrace with gdb. > > Cheers, > Daniel > > On 01.05.19 20:26, Grant Bagdasarian wrote: > > Hello, > > > > I was trying out the auth_ephemeral module inside the xhttp > > event_route in Python KEMI, and discovered a possible bug. > > > > https://pastebin.com/4Ghumfis > > > > The following will result in the stack smash error. Whenever I leave > > out the unixtimestamp the > > module will complain about not being able to convert to an int, which > > make sense, but doesn't lead to a stack smash error and a complete > > crash of Kamailio. > > > > KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). > > > > According to the documentation this module can only be used in > > REQUEST_ROUTE, but I found a presentation online which also used this > > module and that particular function in event_route, which is what I > > need for WebSocket authentication. > > > > Hope this helps. Please let me know if you need more information. > > > > Regards, > > > > Grant > > > > _______________________________________________ > > 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 - May 6-8, 2019 -- www.kamailioworld.com >

2 10

Kamailio Load Balancing
by Gaurav Bmotra 30 May '19

30 May '19

HI i am using kamailio 5.1.2 With Ubuntu 18.04.2 LTS and using Freeswitch as Media server and kamailio as Register server i wan to implement load balance on this plz help thank you Gaurav -- *Regards:* Gaurav Kumar

3 7

usrloc extra attributes and db_mode=0
by Patrick Wakano 29 May '19

29 May '19

Hello list, Hope you all doing well! I am trying to use the extra attributes (xavp_contact) of the usrloc module to save some additional info about the user. I am setting the value before the save() and doing a call to registered() (not lookup()) before trying to access these extra attributes. This works fine with db_mode=3 but does not work in case of db_mode=0. Does anyone knows if it should also work with mode 0? I was expecting it to work but a call to kamcmd ul.dump show nothing related..... I don't want to use the DB for the location purposes because I prefer to have multiple servers using the dmq_usrloc (that works like a charm!) and it conflicts with DB persistence of the location table (when the DMQ replication happens, all servers (sharing the same DB) try to save the same user info in the location table) Thank you, Kind regards, Patrick Wakano

2 6

Statistics help
by Duarte Rocha 29 May '19

29 May '19

Greetings, I'm doing some monitoring of my Kamailio in order to quickly identify system failures and overloads. In order to do that i have scripts that read the output of "kamcmd sl.stats" and identify the rising of "500_replies", "5xx_replies" and "6xx_replies". However, i have rate limiting on my Kamailio, which generates "503" replies when limit is reached, which generates an unnecessary alarm. Is there a way to get more specific stats about the repleis generated by Kamailio? If that isn't possible, will i be safe only monitoring "500" and "6xx" replies? In my experience, when the server has reached a processing or memory limit, the reply is always 500, but i'm not really sure about this. Thank you for your help. Best Regards, Duarte Rocha

2 1

Push proxy and load balancer for asterisk?
by Gerrit Jacobsen 29 May '19

29 May '19

Good morning, I want to realise a push notification proxy to signal incoming calls from asterisk to softphones. Is Kamailio the preferred solution for this? I looked also at Flexisip and Asterisk scripts and was told by a seasoned professional that Kamailio is the preferable solution as its scalable. True? I would be glad if someone could share his experiences. Also can one Kamailio instance be used as an asterisk load balancer and also push notification proxy ? I have been a heavy openser user a long time ago. I used first openser as my core routing engine but then switched to yate. Please excuse my rather broad questions. From my experience with Openser the devil is in the detail. So any hints on where this setup may cause problems would be great. Thanks a lot for your input. Gerry

3 2

Kamailio Lord Balance
by Gaurav Bmotra 29 May '19

29 May '19

hi i m using kamailio 5.1 with Ubuntu 18.04.2 LTS and freeswitch as Media server -- *Regards:* Gaurav Kumar

4 4

RFC: split out old sr api from app_lua
by Daniel-Constantin Mierla 29 May '19

29 May '19

Hello, I am considering to split the app_lua module in order to move the older style API available under Lua 'sr' module in a dedicated module (like app_lua_sr). Later this module can be obsoleted and removed, once we know that everything there is covered by KEMI KSR Lua module. In this way, we make it more clear that KSR is the preferred API to use as well have the app_lua smaller, for those using only KSR and not needing 'sr' API. Therefore I am starting the discussion to see if anyone has something against or other suggestions. I cc-ed sr-dev to make developers aware, but I think the best place for discussion is sr-users, being related to how Kamailio is used, therefore reply only to sr-users. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

RTPEngine / Transcoding question
by Carsten Bock 29 May '19

29 May '19

Hi, I want to implement selective transcoding, e.g. avoiding Transcoding between a HD codec (G722) and a non HD Codec (G711). In case the caller does offer HD codecs, this is fine, e.g. if(sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_offer("codec-transcode=OPUS codec-transcode=G722 codec-transcode=PCMA"); } else { rtpengine_manage("codec-transcode=PCMA"); } Now I have the issue, if the callee only sends me G711, I don't want to offer G722 or Opus to the caller. However, rtpengine_answer() does not seem to accept "codec-strip=G722 codec-strip=OPUS", e.g.: onreply_route() { if(!sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_ manage("codec-strip=OPUS codec-strip=G722"); } } As a result, the SDP always contains the transcoding options, e.g. G722 and OPUS. I always end up in transcoding G722 to G711, which is meaningless in that case. Any ideas on how to solve that? (above examples very simplified) Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

2 5

RTPEngine: rtpengine_manage(), Subsequent requests, Transcoding
by Carsten Bock 29 May '19

29 May '19

Hi, quick question: If understood it correctly in the past (and it worked for me quite well this way), I can simply call rtpengine_manage() for subsequent requests without adding any additional information (e.g. used interfaces, RTP/SRTP translation or anything like that), as this information is stored in the internal hash table of the RTPEngine module during call setup and initial offer/answer. However, this does not seem to apply to the transcoding options. When I just call "rtpengine_manage()" for a Re-INVITE of a transcoded call (in my case AMR-WB to G722), the transcoding options are no longer applied. The forwarded offer does not get the transcoded codecs added (e.g. in my case G722 is not added to the list of AMR/AMR-WB from the phone). Did I misunderstand something or is it a bug? My RTPEngine is still on 7.2, but I will double check with more recent versions today or tomorrow. However, I also found no commit messages relating to this. Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

3 2

Call Routing at IMS
by Iman Mohammadi 29 May '19

29 May '19

Hi Everyone When P-CSCF receives INVITE from client it sends INVITE to S-CSCF then S-CSCF without any look up(checking the user is registered or not)sends INVITE to I-CSCF then I-CSCF sends LIR and receives LIA and sends INVITE to same S-CSCF (in case of one S-CSCF exists).My Question is here why without any look up S-CSCF sends INVITE to I-CSCF??this procedure affects system performance and increases call set up time

2 1

WebRTC bad message (offset: 22)
by Ilie Soltanici 28 May '19

28 May '19

Hello, Can someone help me to discard those messages from kamailio log files? ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS 212.2.161.34:60477#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: VeiXbJClWneIrIyYgM5v3XKYHLA=#015#012Server: WebRTC GW#015#012Content-Length: 0#015#012#015#012> As I understand this is just an HTTP response to the first request, and shouldn't go through config file execution, but I don't understand why it is getting through it. Thank you

1 0

kamailio webrtc 183 early media.
by Zhan Bazarov 28 May '19

28 May '19

Hello! I have setup kamailio+rtpengine+asterisk.. And I have issue with early media for WEBrtc endpoints... As I understood right, for this case I need to use rel100(PRACK) for sending early media... But I didn't find any example for solve this issue... Can you help me with it? It would be greatly appreciated!

1 0

mem leak on 5.1.2?
by Andrew Chen 27 May '19

27 May '19

Hey guys, Need some help with possible run away memory issue on my 5.1.2 Kamailio. I noticed I'm getting this memory error from kam logs: May 24 13:47:52 ashintgtpsg51 /usr/sbin/kamailio[1857]: WARNING: http_client [functions.c:298]: curL_query_url(): Curl library out of memory (url:https://<some url>) my -m on kamailio startup is currently set to 64 but there is only 1 call going through. In our check_mk, I noticed the memory usage is ramping up: [image: image.png] What's the cause of this? -- Andy Chen Sr. Telephony Lead Engineer 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.*

4 8

Getting the following error when sending INVITE request for calling
by Mohit Chawla 27 May '19

27 May '19

Hi , I am getting the following error in trying to send INVITE message from SIP client. {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: timestamp=1558902843;method=INVITE;from_tag=69588468;to_tag=ebd29a60;call_id=4_Do2VIMWLR6Wf5ubiBxgg..;code=200;reason=OK;src_user=1016;src_domain=172.25.22.23;src_ip=172.25.22.40;dst_ouser=1015;dst_user=1015;dst_domain=172.25.22.40 May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert into database May 27 02:04:10 epc-srv-03 /usr/sbin/kamailio[17229]: {2 3 BYE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: *Can anyone help me understand the reason for this error.* *I am attaching the used kamailio.cfg file in this message.* *Thanks and Regards,* *Mohit* -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 6

remove_hf() leaves junk in the headers?
by Oded Arbel 27 May '19

27 May '19

I'm trying to rewrite the `From` header in an invite going out to an external system (*), by calling remove_hf("From"); followed by append_hf("From: <some value>\r\n"); but the result is an invalid SIP message, because apparently after `remove_hf()` finishes, it leaves the content of $fU in place of the `From` header - just the raw text, without a header name or even a "\r\n" - and that eats the next header line and breaks the INVITE. Here is how the INVITE originally looks (without either `remove_hf()` or `append_hf()`): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=649.f0b> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK6188.ff5c4fd855bd2970fdf32d4807b114c2.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK581897a4;rport=5066 Max-Forwards: 69 From: <sip:972515340014@example.com:5066>;tag=as0bcee217 To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 722299dc15aad53d28172d433e8665f4@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:42:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ad199638-4df3-4aef-81f1-1b50581032a7 Content-Type: application/sdp Content-Length: 280 v=0 o=root 817636675 817636675 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39064 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39065 a=ptime:20 ---8<--- and here is another call with the same setup, with just `remove_hf("From");` being invoked (without calling `append_hf()` - this is just for reproduction): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=f45.91b1> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK024d.2690ebf7a809edca2208e6b2561ccf9e.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK79ae3073;rport=5066 Max-Forwards: 69 972515340014To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 7b20ba5a549f6f886c22985146765736@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:44:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ffbb4ac1-3a0f-4dd6-9ce9-7ad24361081a Content-Type: application/sdp Content-Length: 280 v=0 o=root 509740923 509740923 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39102 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39103 a=ptime:20 ---8<--- As you can see, the `From` header is indeed gone, but it left the $fU just before the `To` header. WTF? I'm running Kamailio 5.2.2 built locally. Notes: *) the problem I'm trying to solve is that I can't get my Asterisk installation to generate the `From` header I want, the reason is out of scope of this question, so I'm sending custom SIP headers over to Kamaiio and I want to rewrite the header in Kamailio to the "correct" value. -- Oded Arbel

6 8

PRE-PAY update timer
by Steve Bucklin 25 May '19

25 May '19

Hello all (again), I am looking at a 'good' method to start a call with "chunks" of time allocated. I wish the initial call to check a credit, and if credit is good, allow a small amount of time (say 30 seconds). I have enabled DIALOG and have looked at CNXCC, but am confused! When the initial 30 seconds runs out, I want to authorise another 30 seconds, etc, etc, etc. I have started to play and set a dialog timer that expires to a route, and then can 'return' allowing the call to continue - BUT was not able to reset the dialog timer to trip again after another 30 seconds. I tried to 'set_max_time' on cnxcc, but this did not seem to trip the event route upon timeout? Does anyone have a pointer to any examples? I also am unsure of how to update the timer and keep a call in play! Steve

5 4

No Private memory left using usrloc with db_redis in mode 3
by José Seabra 24 May '19

24 May '19

Hello there, i have stored ~200 000 registrations contacts in redis database and when kamailio starts looking for expired contacts in redis then it starts giving these errors: WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires' *WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7ff80922b010, 3055920); Free fragment not found!* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7ff80922b010, 3055920) called from core: db_res.c: db_allocate_rows(183), module: core; Free fragment not found!* *ERROR: PR_LOG: 123: <core> [db_res.c:185]: db_allocate_rows(): no private memory left* *ERROR: PR_LOG: 123: db_redis [redis_dbase.c:1115]: db_redis_perform_query(): Failed to allocate memory for rows * I have configured the private memory to 128 MB but seems that isn't enough to handle with all these contacts. Is the only option here increase the private memory or there is any other kind of configuration(in redis or kamailio) that i can make to avoid this? Best Regards José Seabra -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Kamailio IMS installation from pacakges
by Mohit Chawla 24 May '19

24 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit From: Mohit Chawla [mailto:mohit.chawla@azcom.in] Sent: Wednesday, May 22, 2019 5:57 PM To: 'sr-users(a)lists.kamailio.org' Subject: Kamailio IMS installation from pacakges Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 3

Use of kamailio as IMS server to integrate with LTE Core Network
by Mohit Chawla 24 May '19

24 May '19

Hi , We have successfully installed the kamailio IMS server on centOS 7 using the following link (http://www.kamailio.org/wiki/tutorials/ims/installation-howto) with appropriate changes for command in CentOS 7. We are currently trying to integrate the PCSCF module with LTE Core network , but I don't have any idea how to do this. Neither I am able to find any tutorials which I can use to understand how kamailio works. Can anyone share any reference document to help me understand the kamailio working in my use case. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

listen (bind) to variable ip and port
by Roiss-Weiss, Roman 23 May '19

23 May '19

Hi guys, is it somehow possible to bind to a variable ip-address and port? e.g. getting the ip and port from a database? Kind regards, Roman

3 2

Kamailio CDRs for calls interrupted without BYE
by Ildar Mukhametzhanov 23 May '19

23 May '19

Hello! I am using dialog+acc modules to write CDRs in kamailio. Everything works great except one test. During active call I switched off both UACs preventing them to send BYE. Rtpengine reported about call interruption, but kamailio thinks that this call is in-progress for a long time. What is the right way to make kamailio know about calls interrupted without BYE. Thanks. -- Sincerely, *Ildar Mukhametzhanov* Head of software development VoiceON Telecom Services GmbH im(a)voice-on.net <mailto:im@voice-on.net> t. +49 30 398 20 500-77

3 8

warning logs - DB_Redis and usrloc in mode 3
by José Seabra 22 May '19

22 May '19

Hello there, I'm testing the db_redis with usrloc module in mode 3 and i have been noticing that every time that kamailio checks if there is contacts expired then it logs the following warning messages: *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1274]: db_redis_perform_delete(): performing full table scan on table 'location' while performing delete* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 1 is 'expires'* I'm not sure what is the rational behind of this, but having these logs being written when we have log level equal 2 in production is something unnecessary, so i'm asking if it shouldn't be better we have it as debug instead of warning? or do you forecast any performance issues using usrloc module in mode 3 with db_redis? Thank you for your support Cheers José -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 22 May '19

22 May '19

Benoit, Alex, I appreciate the feedback, confirming my assumptions. Good news is the Vendor finally responded positively and agreed to add in configuration option to route on INVITE R-URI in next SDK release, expected later this year. In the meantime I added 'uac_replace_to' function to my proxy to re-write To: header with R-URI, so far so good. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope > > The To header is a cosmetic, purely logical commentary on the intended ultimate destination of the call, and no routing should ever be done on it. The RURI of the INVITE is the appropriate vehicle for that. > > When dealing with proxies, however, one will encounter from time to time the grim reality that there are user agents which expect the To URI and the INVITE RURI to be in alignment. For that scenario, you can use ‘uac’ module’s stateful rewriting functionally (uac_replace_to()) to modify To with Kamailio in a way that displays fidelity and esteem for protocol formalities. > > — > Sent from mobile, with due apologies for brevity and errors. > > > On May 21, 2019, at 9:32 AM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > > > > Hey Folks, > > > > I could use some feedback to see if my mind is right on this topic.I'm > > in a discussion with a software Vendor (respectfully unnamed) with > > terminal software routing on To: header info. It's my contention any > > modern SIP software should use INVITE header to retrieve DID info and > > route from there. > > > > My reasoning is due to the To: header should contain the original > > intended DID, Extension, Name, AoR or whatever else the original SIP > > transaction wanted to contact, but being in a multi-carrier, > > multi-hop, call forwarding environment, the INVITE header contains the > > hop-to-hop true intention of where the call should route at any given > > transaction. So in the case of terminal software the To: header could > > likely be irrelevant where as the INVITE header is accurate. > > > > My most recent response from the Vendor suggest I put a SIP Proxy in > > front of the terminal software and transform the To: header to > > whatever I need for this application. Although I can do this, I'm > > concerned with breaking RFC for CANCEL and BYE transaction sent back > > from the terminal software that may not be recognized by upstream > > origination carriers. > > > > Plus modifying To: header is widely frowned upon by most folks. > > > > Thanks. > > > > JR

1 0

Kamailio v5.2.3 Released
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, Kamailio SIP Server v5.2.3 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.3. For more details about version 5.2.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/05/kamailio-v5-2-3-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.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio IMS installation from pacakges
by Mohit Chawla 22 May '19

22 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

Releasing Kamailio v5.2.3
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, I am considering to release Kamailo v5.2.3 out of latest git branch 5.2 next week, likely on Wednesday, May 22. As usual, if you are aware of issues not reported to the github bug tracker, fill a report there as soon as possible to give a chance to be analyzed and fixed. Also, if there was a fix for an issue that you were reporting, check to see if it was backported to be sure it is not forgotten. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

remove_hf inside of onreply_route
by Володимир Іванець 21 May '19

21 May '19

Hello! I was wondering if someone is using remove_hf in onreply_route and if yes on which Kamailio version? I'm using Kamailio v5.2.1 and as soon as I add remove_hf function to onreply_route Kamailio will fail to start with following error messages: ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fc1dc044010, 27120); Free fragment not found! ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fc1dc044010, 27120) called from core: core/io_wait.c: init_io_wait(524), module: core; Free fragment not found! Thank you for any suggestions!

2 2

Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 21 May '19

21 May '19

Hey Folks, I could use some feedback to see if my mind is right on this topic.I'm in a discussion with a software Vendor (respectfully unnamed) with terminal software routing on To: header info. It's my contention any modern SIP software should use INVITE header to retrieve DID info and route from there. My reasoning is due to the To: header should contain the original intended DID, Extension, Name, AoR or whatever else the original SIP transaction wanted to contact, but being in a multi-carrier, multi-hop, call forwarding environment, the INVITE header contains the hop-to-hop true intention of where the call should route at any given transaction. So in the case of terminal software the To: header could likely be irrelevant where as the INVITE header is accurate. My most recent response from the Vendor suggest I put a SIP Proxy in front of the terminal software and transform the To: header to whatever I need for this application. Although I can do this, I'm concerned with breaking RFC for CANCEL and BYE transaction sent back from the terminal software that may not be recognized by upstream origination carriers. Plus modifying To: header is widely frowned upon by most folks. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

4 3

$snd(proto) mapping?
by Andrew Chen 21 May '19

21 May '19

Hey guys, Is there a numeric mapping for $snd(proto)? Thanks -- Andy Chen Sr. Telephony Lead Engineer 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 1

How to get started (configuring an interconnection)
by Benoit Panizzon 21 May '19

21 May '19

Dear Kamailio Users I might need a bit help with starting. We are a TSP and at the moment we use a commercial carrier grade voice switch, which is very VERY inflexible and where bugs we find either don't get fixed, or only with a large delay. As everywhere else we embrace opensource and enjoy the possibility to quickly fix bugs ourselves (we have a software development department) we are evaluating Kamailio. Basically, if we have a centralized SIP routing engine, which can record CDRS for Billing and do some Number Translations (lookup on external service or database) for (location based emergency numbers, call forwarding services, ported numbers, call spam blacklisting) we could do way more than our actual $$$ carrier switch is capable of. Kamailio sounds like perfect for this task. First step: Get a machine up and running with Kamailio and Siremis, done, that was easy. Next: Connect his SIP wise via what I would call a 'SIP carrier interconnecton', ip based authenticated trunk between our carrier switch and Kamailio so we can route some number ranges from our carrier switch to Kamailio and then continue to test with subscribers, subscriber trunks and all else we need on Kamailio. I'm quite fluent in Asterisk (our Voicemail and Announcement Services are based on Asterisk) and SIP and have come across many different PBX, but after reading parts of the documentation, and trying myself to understand how Kamailio talks to other SIP endpoints I am a bit at a loss. I would have expected that I would configure our carrier switch as 'remote SIP gateway' or whatever it would be called. But I start fearing, Kamilio works completely differently. So I might need a bit of pointing the right direction to get started. I did try with the 'CarrierRoute Management' or 'Dispatcher' functions, but I only find very little documentation on this. So how do I get started? How do I, for starters, tell Kamailio what the Hostname or IP of my Carrier switch is and how do I set a 'default' call route to this? Kind regards -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Storing an integer value into AVP from REDIS string
by Stephen Bucklin 21 May '19

21 May '19

Hi, Any quick thoughts. I am using a REDIS "GET" to retrieve a JSON obj as a string from REDIS and then the JANSSON get value to extract a key value from the string and pop it into an AVP ($avp(val)) and wish to ensure the value is an integer in the AVP. New to this, be kind :-) Steve

4 6

How to define IP Trunk in kamailio
by Mary Honey 21 May '19

21 May '19

Hello everyone, I install and setup an IMS Network by kamailio-5.2.2 . I have p-cscf, i-cscf and s-cscf in three different systems . my Hss is Fokus. Registration and call between subscribers of network work successfully . now I want to define IP Trunk in my network which can pass external calls through my IMS network. I don't know how to define IP Trunk and changes in which element(pcscf, icscf or scscf) should be done.

2 3

Rewrite contact header
by R, Rajkumar (Raj) 21 May '19

21 May '19

Hello, We are using alias routing from NATHelper module and I've a case where the alias part 'alias=ip~port~transport' needs to be removed from the contact header in 2xx message. There's contact header reference "$ct" pseudo variable but it is read only. Is there any approach available to achieve this as I haven't found anything in the forum archive search so far. thanks, raj

2 2

Unable to CDRs on "Busy Here 486" from remote side and save start and end time on successful call in table
by Shahid Mehmood 21 May '19

21 May '19

Hello, i want to save the status each call into the database. but failed to 1. record CDRs for "Busy Here 486/ service unavailable 503 " from remote side in database. 2. on successful call unable to record total time of call in database. Version of Kamailio Is *version: kamailio 4.4.7 (x86_64/linux) 35d8abflags: 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_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: 35d8ab* *Also Attached Config File* *Thanks*

1 0

kamailio Early IMS security
by eyas barhouk 20 May '19

20 May '19

hello community Does kamailio support Early IMS security authentication schema to be able to integrate with Huawei HSS ?? and if it's does how to configure it? B.R Eyas

1 0

Plans for next major release - v5.3.0
by Daniel-Constantin Mierla 20 May '19

20 May '19

Hello, a while ago, during the last IRC devel meeting, it was proposed to either do the next major release (5.3.0) before the usual summer holidays or leave it for autumn. Given that time is approaching, we have to decide. If it is going to happen before the summer holidays, probably we should freeze the development in 2-3 weeks, then do 3-4 weeks of testing and release during the first part of July. It would be a shorter development cycle, so I do not expect much needs for testing. Only a few new modules and plenty of work to consolidate and enhance the existing code. Any preference for one of the two options? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 2

Kamailio usrloc mode 3 using mongodb
by José Seabra 20 May '19

20 May '19

Hello there, I'm testing usrloc module with mongodb and i have a couple of questions to do. What i would like to know/have confirmation is if the fields that are specified in the following link https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-location-… can be used to create indexes in mongodb? I'm noticing that when a register is updated/deleted or expires kamailio prints the following INFO logs: *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (methods) type id (10), use DB1_STRING as default* *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (instance) type id (10), use DB1_STRING as default* These fields (*methods and **instance*) are stored in mongodb document with a null string, e.g: *{* * "_id" : ObjectId("5cdc775a6472740055289a11"),* * "username" : "x1000",* * "contact" : "sip:x1000@192.168.1.1:5674;transport=UDP",* * "expires" : ISODate("2019-05-15T20:33:26Z"),* * "q" : -1,* * "callid" : "sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61 <sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61>",* * "cseq" : 2,* * "flags" : 0,* * "cflags" : 0,* * "user_agent" : "sipp",* * "received" : "sip:191.30.10.23:8888 <http://191.30.10.23:8888>",* * "path" : "<sip:172.31.2.61:5060;received=sip:191.30.10.23:8888;lr>",* * "socket" : "udp:172.31.2.11:5060 <http://172.31.2.11:5060>",* * "methods" : null,* * "last_modified" : ISODate("2019-05-15T20:32:26Z"),* * "ruid" : "uloc-15-5cdc7756-55-1",* * "instance" : null,* * "reg_id" : 0,* * "server_id" : 21,* * "connection_id" : -1,* * "keepalive" : 0,* * "partition" : 0* *}* Should i be concerned about that log msg? how can i avoid it? Thanks Regards -- Cumprimentos José Seabra

2 5

BYE relays Digest and others in Proxy-Authorization header
by Sergiu Pojoga 19 May '19

19 May '19

Hi ppl, Noticed BYE requests relayed from caller to callee have the *Proxy-Authorization* header including *Digest*, *nonce* and *response*. Is that by SIP definition or am I doing something wrong? My concern is the callee gets to see caller's authentication credentials. Thanks.

4 6

Centos7 Packages
by Matt Lamer 18 May '19

18 May '19

I am trying to install the kam server on centos7. When running the install I get the following: Install 5 Packages (+2 Dependent packages) Total size: 14 M Total download size: 14 M Installed size: 63 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-debuginfo-5.2.2-7.el7 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Anyone else having this issue?

2 1

"If you're struggling with your Tele callers to improve sales? you're not alone.
by Prakash 17 May '19

17 May '19

http://www.doocti.com +919790776655 Doocti Next Generation CTI for Smarter Sales Teams http://www.doocti.com Dear Customer, Greeting from Tevatel. Hope this mail finds you well. We are happy to announce the launch of our new product Doocti - Call center connector for crm What is Doocti? Doocti is a Computer telephony integration platform that can integrate your communication devices with your Helpdesk, CRM, and Other Key Tools. Increase agent productivity, improve customer experience and enable personalized service through deep customer insights at the point of interaction. Features of Doocti Single click Dial On-Screen Customer Profile Build on the Customer Profile Efficient Call Disposition Auto Dial Sticky Agent Feedback from the clients Reduce call time by 10-20 seconds per call Instantly retrieves and screen-pops customer related information Provides a better and more immediate response to the customers Offers a more personalized experience to the caller Dials out automatically and faster with a single mouse click Handles higher call volumes Increases first call resolution Intergrated with Currently we have associated with more than 20 brands. Curious to see what's under the hood? Enjoy a detailed look at the Doocti experience in a demo guided by a member of our team. Get a meeting on the calendar today. mailto:prakash.n@tevatel.com?subject=Need More Information about Doocti Request for a Demo [ mailto:prakash.n@tevatel.com?subject=Need More Information about Doocti ] Communications Simplified Contact Us: +919790776655 | prakash.n(a)tevatel.com No:76, Swamibakthan Street, Nammalwarpet, Chennai - 600012. www.tevatel.com | www.doocti.com http://viewstripo.email/?utm_source=templates&utm_medium=email&utm_campaign… ---------------------------------------------------------------------- This email was sent by prakash.n(a)tevatel.com to sr-users(a)lists.sip-router.org Not interested?Unsubscribe - https://tihk.maillist-manage.com/ua/optout?od=27218d28c96aa859ed0ac1287be0e… Update profile - https://tihk.maillist-manage.com/ua/upc?upd=164d3589027540e5&r=164d35890275… Asterfone technologies pvt ltd | Swamibakthan St, Thiruvalluvar Nagar, Nammalwarpet, Ayanavaram, Chennai, Tamil Nadu 600010 Our Privacy Policy [ ] and Terms of Use. [ ]

2 1

rtpengine sessions on timeout not freeing up RTP ports
by Istvan Mogyorosi 17 May '19

17 May '19

Hello Another rtpengine related question after having successfully implemented a dispatcher configuration. Call flow is ok, everything works well until I tried to increase call volume. Kamailio log showing "proxy replied with error: Ran out of ports" In the beginning both channels are up, then I can see a teardown on the incoming channel loosing the ability to collect DTMF. Using the CLI tool I have statistics showing that none (!) of the calls are "regular terminated", they are on timeout. After experimenting with handling BYE, I ended up could getting a "regular terminated" call by using route[RELAY] .... if (is_method("BYE")) { rtpengine_delete(); } and then I can see some good things! Received command 'delete' from 127.0.0.1:58538 Scheduling deletion of call branch 'as65b95845' (via-branch '') in 5 seconds Replying to 'delete' from 127.0.0.1:58538 (elapsed time 0.000216 sec) Please correct me if I'm doing wrong this way! cheers Istvan

1 0

RFC: Kamailio command line parameters
by Daniel-Constantin Mierla 17 May '19

17 May '19

Hello, a while ago I added support for long format parameter names for Kamailio command line. That's opening the door for more options to be set via command line arguments and I discovered that especially in a containerized environment, it is a trend to start applications using more command line parameters, than relying on config files. The current command line parameters are listed by: kamailio -h Use the master branch for it, because has many more than 5.2.x. Anyhow, the purpose of this message is to see if anyone else considers useful to add other command line parameters, and if yes, which ones. A typical candidate will be a global parameter that you have to change when relocating to/spinning up another Kamailio environment and you need to change the kamailio.cfg. Those that generate the interest of many people, can be easily added now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Testing the KEMI interpreters for functions with 4 and 5 parameters
by Daniel-Constantin Mierla 17 May '19

17 May '19

Hello, because there were many variants not handled for KEMI functions with 4 and 5 parameters, I made a script to generate the code in order to be sure none is missing. The script was pushed in Kamailio repo at: * https://github.com/kamailio/kamailio/tree/master/misc/tools/kemi You don't have to use it, I listed it for reference, just in case the code needs adjustments in the future. If anyone is using a KEMI interpreter and has functions exported by Kamailio with 4 or 5 parameters, it would be appreciated if you can run some tests with master branch and report the results. All kemi interpreter modules were updated, so it doesn't matter which one you test. If the results are good, then the patches will be backported to branch 5.2, so there will be no case of unhanded functions with these numbers of parameters. The cases with up to 3 parameters were not affected. For 6 params, still only string values are handled. The other variants can be added, but I think they are not needed for the moment, so I tried to keep it small for the moment. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio World thanks
by Daniel Pocock 16 May '19

16 May '19

Hi Daniel and Kamailio team, Well done on another successful Kamailio World in 2019. As usual, it was a great opportunity to meet people and learn new things. Regards, Daniel

2 1

siputils add_uri_param not available in KEMI python3
by Patrick Murphy 15 May '19

15 May '19

Hello, I am using the following version of Kamailio and using app_python3 with KEMI but I cannot find any exported function of add_uri_param from the siputils module. Any idea how I can perform: KSR.siputils.add_uri_param("name=value") ? version: kamailio 5.2.2 (x86_64/linux) 9260a2 Cheers, Pat.

1 1

Using a single virtual domain in a multi-domain setup
by Rhys Hanrahan 15 May '19

15 May '19

Hi All, I am trying to setup Kamailio to allow me to have multiple different domains (and IPs) treated as one. I am using the domain module for this. As an example I have: * A handset registered to Kamailio by IP address. * A SIP trunk between Asterisk and Kamilio using an FQDN that uses SRV records for redundancy on Asterisk's side. I have added all domains and IPs under the same "domain identifier". It seems the "old" domain module now uid_domain had this capability of a single virtual domain, but is this possible with the more current domain.so module? From what I've said the uid_* modules older modules from the SIP Router project, and should be replaced by more recent Kamailio modules? - https://www.kamailio.org/docs/modules/5.0.x/modules/uid_domain.html VS. - https://kamailio.org/docs/modules/5.1.x/modules/domain.html MariaDB [kamailio]> select * from domain; +----+------------------------------------+------------+---------------------+ | id | domain | did | last_modified | +----+------------------------------------+------------+---------------------+ | 1 | 12.34.56.53 | vocalpoint | 2000-01-01 00:00:01 | | 2 | 23.34.56.54 | vocalpoint | 2000-01-01 00:00:01 | | 3 | sbc.foo | vocalpoint | 2000-01-01 00:00:01 | | 4 | sbc.nsw.foo | vocalpoint | 2000-01-01 00:00:01 | | 5 | sbc.vic.foo | vocalpoint | 2000-01-01 00:00:01 | | 6 | sbc.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 7 | sbc.nsw.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 8 | sbc.vic.foo2 | vocalpoint | 2000-01-01 00:00:01 | +----+------------------------------------+------------+---------------------+ 8 rows in set (0.000 sec) I managed to get registrations working by hardcoding a static realm in the config, but when when usrloc does a lookup on a handset, Kamailio can't find user(a)sbc.nsw.foo and Kamailio returns 404. It looks like this is because in the locations table it is listed with a domain of 12.34.56.53 (because the handset is registered by IP). How should this work normally? Or is it even possible? Thanks, Rhys.

2 4

Sip proxy for Asterisk with RFC3966 and 100Rel
by paolo.visnoviz＠vipcomputers.it 15 May '19

15 May '19

Dear Sirs, a telco provider give us a trunk that require 100rel and TEL RFC 3966. Usually we use Asterisk. Now Asterisk 16 with PJSIP support 100Rel, but not RFC3966, and they don't actually works to implement it. Asterisk chan_sip, instead, support RFC3966, but not 100Rel and is not actively developed on by Digium or Sangoma. So I'm stuck. Now, do you think it was possible to use Kamailio to act like a proxy sip but implementing RFC3966 for incoming and outgoing calls and RFC3966 and 100Rel for outgoing calls? And it is possible to connect all to an Asterisk server? Thank you in advance Best regards. Paolo Visnoviz

3 9

Possible memory leak in siptrace module in kamailio 5.1/5.2
by davy van de moere 15 May '19

15 May '19

Dear, I'm using siptrace within a python handler in the KEMI framework. KSR.siptrace.sip_trace_dst("sip:<myhomerip>:9060") I notice an ever increasing usage of memory, until the system runs out of memory. (definition of a memory leak :) ). Looking at the memory information kamailio is giving me I notice this: May 14 05:49:48 server kamailio[36452]: NOTICE: PYT {INVITE} {1769595029-1513210001-1955500877} {1} : qm_sums: count= 79286 size= 5812048 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) May 14 05:51:17 server kamailio[36452]: NOTICE: PYT {REGISTER} { 4_1419955802(a)192.168.0.153} {2582} : qm_sums: count= 79335 size= 5814840 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) I notice this on kamailio 5.1.8 (from a debian repo), and on kamailio 5.2.2 (from source). I can reproduce the problem very easily, and I can apply patches in the source to try out things. Could I be wrongly using siptrace? Can I give you more debugging info? Kind regards, Davy

3 4

define a DBURL with SLQOPS fails
by David Villasmil 14 May '19

14 May '19

Hello guys, So, i'm defining: DBURL "mysql://user:password@dbhost/dbname" and later on... modparam("sqlops","sqlcon","cb=>"DBURL) and this fails miserably.. What do do on this case? Thanks all! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 12

Npdi question
by Duarte Rocha 14 May '19

14 May '19

Greetings, Regarding portability issues, is there a difference between "npdi" and "npdi=yes" ? Are there any more possible values besides "yes" and no value? Best Regards, Duarte Rocha

1 0

sdp_remove_codecs_by_id does not remove 101
by Juha Heinanen 14 May '19

14 May '19

I tried to use sdp_remove_codecs_by_id("101") to remove "codec" 101 (dtmf) from sdp, but nothing happened. It is still there: m=audio 31128 UDP/TLS/RTP/SAVPF 101 96. c=IN IP4 192.168.43.88 a=label:1. a=rtcp-rsize. a=ssrc:2880178656 cname:sip:test@test.tutpro.com. a=rtpmap:101 telephone-event/8000. a=rtpmap:96 opus/48000/2. a=fmtp:101 0-15. Is that by design or a bug? If not a bug, is there some other means to remove 101? -- Juha

2 2

sip trunk with kamailio
by arish haque 14 May '19

14 May '19

Hi, I am trying the SIP trunk from kamailio with following configurations - cfg_get.pstn.gw_ip="10.0.x.y route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(!($rU=~"^(\+|00)918884978235")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } I am able to see kamailio sending INVITE to SIP Provider but no response back. Any other, configuration is required which I'm missing. I have attached the tcpdump snapshot of the above scenario. Thanks, Arish

2 1

Theoritical Anycast Setup
by Daniel Oakes 14 May '19

14 May '19

Hi there, Long time user, 1st time poster - apologies if someone has posted this previously. We have a product from a vendor that utilizes Kamailio as a SIP router - but we're wanting to extend it a little bit by the use of anycast (we're a small ISP). Basically the scenario is that we want to utilize Quagga via BGP to anycast an address, which addresses a glaring load balancing issue with have with people who use only A records. Problem is anycast doesn't work so well in this scenario, just by adding the Anycast as a listen address on the proxy. What we'd want to do is rewrite the SIP so that all responses come back out saying to use the 'real ip'. So I have two variables - say REAL_IP_ADDRESS and ANYCAST_IP_ADDRESS - request will come in via ANYCAST, but I want all responses from then on to be via REAL_IP_ADDRESS. We have six Kamailio boxes in various geographical locations, so the idea is they hit one based on closest, and we rewrite it so they end up with the real ip address to talk to. Anyone done something similar? I'm guessing possibly, but I can't find anything specific enough for me to use. Help appreciated! Regards, Daniel

2 1

No audio after IP address change
by Chris Scheid 11 May '19

11 May '19

Hi, we are using rtpengine with Kamailio using DTLS-SRTP and SRTP making audio calls from two Android devices. Regular audio calls work fine but we have issues with the following scenario: 1. Device 1 (72.66.83.8, on WiFI) calls Device 2 (174.204.14.29). SIP server has IP 209.182.216.71. 2. Device 2 switches networks from WiFI to 3G (new IP: 97.47.70.223). Re-invite flow completes. 3. Any audio from Device 2 cannot be heard on Device 1. But audio from the Device1 can be heard on the Device 2. In the rtpengine stats I see that there were some packets sent from the SIP server to the Device 1 after network change: [1557513316.000308] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000318] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20052 <> 97.47.70.223:4667 , SSRC 77c38349, 55 p, 3410 b, 0 e, 32 ts [1557513316.000325] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20053 <> 97.47.70.223:4640 (RTCP), SSRC 77c38349, 12 p, 1120 b, However, in my UDP trace, I don't see any communication between the SIP server and the Device 1 after network change (IP 97.47.70.223). This seems to be the reason why we don't get audio on Device 1. I'm looking for direction on how to troubleshoot this issue. I'm not so familiar with rtpengine. Any assistance greatly appreciated. Thanks! RTP Engine Config: [rtpengine] #####table = 0 listen-ng = 127.0.0.1:5050 timeout = 60 silent-timeout = 3600 tos = 184 port-min = 20000 port-max = 60000 interface = 209.182.216.71 log-level = 7 log-stderr = false log-facility = local0 START CALL --------------------------------------------------------------------------------------------------------------------- [1557513231.973294] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'offer' from 127.0.0.1:45431 [1557513231.973357] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'offer' from 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502030 3766502030 IN IP4 192.168.1.200 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5095 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 192.168.1.200 b=TIAS:64000 a=rtcp:5096 IN IP4 192.168.1.200 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2009301833 cname:32ababda5d2c847b a=setup:actpass a=fingerprint:SHA-256 4F:69:B6:89:51:98:6C:A4:63:CF:F7:CD:22:D2:BE:FD:6E:3F:F4:2A:8B:39:AD:E4:C0:8D:AE:BD:A3:70:12:49 ", "ICE": "remove", "DTLS": "passive", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9 799-0557b76cc340", "received-from": [ "IP4", "72.66.83.8" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "command": "offer" } [1557513231.973444] NOTICE: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating new call [1557513231.973496] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Default sink codec is speex/16000 [1557513231.973503] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for iLBC/8000 [1557513231.973508] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/16000 [1557513231.973512] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/16000 [1557513231.973515] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/8000 [1557513231.973519] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/8000 [1557513231.973523] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/32000 [1557513231.973526] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/32000 [1557513231.973530] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for GSM/8000 [1557513231.973534] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec GSM/8000 [1557513231.973538] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMU/8000 [1557513231.973542] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMU/8000 [1557513231.973545] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMA/8000 [1557513231.973549] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513231.973545] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMA/8000 [1557513231.973549] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513231.973552] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for G722/8000 [1557513231.973556] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec G722/8000 [1557513231.973560] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for telephone-event/8000 [1557513231.974291] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 192.168.1.200:5095 [1557513231.974561] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 192.168.1.200:5096 [1557513231.974874] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: offer time = 0.001488 sec [1557513231.974882] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'offer' from 127.0.0.1:45431 [1557513231.974900] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'offer' to 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502030 3766502030 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20032 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2009301833 cname:32ababda5d2c847b a=rtpmap:104 iLBC/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20033 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:G9MvyRPBAVFsrMypvr1EAunZiX/vc04sXEq4mXvl a=setup:actpass a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513232.077386] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Processing incoming DTLS packet [1557513232.078219] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Sending DTLS packet [1557513232.189220] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Processing incoming DTLS packet [1557513232.189868] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: DTLS handshake successful [1557513232.189899] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: DTLS-SRTP successfully negotiated [1557513232.189905] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: SRTP keys, incoming: [1557513232.189914] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: --- AES_CM_128_HMAC_SHA1_80 key q67B3h3GP/dhbQeKsgBnvA== salt EcoNJ6Hqj0lRtvZ8Z+Y= [1557513232.189919] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: SRTP keys, outgoing: [1557513232.189923] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: --- AES_CM_128_HMAC_SHA1_80 key 0nTo2gy4XJ2GXlroyyH6nA== salt fbIfUIa4IPiYLvn5CkI= [1557513232.189929] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Sending DTLS packet confirmed [1557513243.648460] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SRTCP output wanted, but no crypto suite was negotiated [1557513243.648486] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513243.648496] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 9fda818d: FL 205 TL 6486816 HSR 613804890 J 3100337071 LSR 1419186059 DLSR 531075031 [1557513243.648505] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513243.649142] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513243.649158] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about d4b1278d: FL 23 TL 2271802 HSR 278642673 J 2408385537 LSR 3 653820601 DLSR 1701759944 [1557513243.649169] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513243.777813] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'answer' from 127.0.0.1:47861 [1557513243.777860] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'answer' from 127.0.0.1:47861: { "sdp": "v=0 o=- 3766502031 3766502032 IN IP4 100.81.62.76 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5079 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 100.81.62.76 b=TIAS:64000 a=rtcp:5080 IN IP4 100.81.62.76 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2006816651 cname:1c0965113e9b1046 a=setup:active a=fingerprint:SHA-256 AD:2D:0F:A6:11:DD:75:CC:3C:50:3A:9A:F0:29:20:8D:95:BC:3E:2A:A0:34:F6:67:14:A7:87:EB:8D:66:77:B1 ", "ICE": "remove", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "a nswer" } [1557513243.777998] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for iLBC/8000 [1557513243.778010] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for telephone-event/8000 [1557513243.778088] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 100.81.62.76:5079 [1557513243.778117] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: DTLS: Peer certificate accepted [1557513243.778123] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 100.81.62.76:5080 [1557513243.778166] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: answer time = 0.000267 sec [1557513243.778175] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'answer' from 127.0.0.1:47861 [1557513243.778187] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'answer' to 127.0.0.1:47861: { "sdp": "v=0 o=- 3766502031 3766502032 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20052 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2006816651 cname:1c0965113e9b1046 a=rtpmap:104 iLBC/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20053 a=setup:passive a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513243.864786] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513243.865568] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513243.912858] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513243.913111] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS: Peer certificate accepted [1557513243.913627] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS handshake successful [1557513243.913668] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS-SRTP successfully negotiated [1557513243.913675] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, incoming: [1557513243.913682] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key +C2ck1NPvsfHYoKflimqaA== salt xM4aFoL60/ePpmQJsT8= [1557513243.913686] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, outgoing: [1557513243.913689] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key VvTNUyescFh+vNj16/fePw== salt m3EuTJjAesJHaSXfqKU= [1557513243.913694] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513244.475362] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513244.475405] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 3873048209 PC 2516599107 OC 673505029 NTP TS 1161195675/4 008032614=1161195675.933193 [1557513244.475412] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about bb94938e: FL 243 TL 11665253 HSR 1302964504 J 2830082235 LS R 1964993324 DLSR 1625750139 [1557513244.475422] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513244.492749] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513244.492775] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513244.492791] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2981125880 PC 4052113656 OC 2653921327 NTP TS 4076801750/ 1160081551=4076801750.270103 [1557513244.492797] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 3813d27e: FL 75 TL 11393233 HSR 1719559869 J 944495881 LSR 790240102 DLSR 447292567 [1557513244.492804] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513247.217290] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Confirmed peer address as 174.204.14.29:10523 [1557513248.958592] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513248.958638] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 3148803183 PC 137367242 OC 1506652935 NTP TS 3606800652/1 621372750=3606800652.377505 [1557513248.958647] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 468fa6ae: FL 57 TL 15596558 HSR 3764947872 J 3769875796 LSR 4259175195 DLSR 3903464346 [1557513248.958657] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513248.958667] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Confirmed peer address as 174.204.14.29:10496 i[1557513249.259063] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513249.259108] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 3424021083 PC 3823893658 OC 584056375 NTP TS 3244065141/3 391763074=3244065141.789706 [1557513249.259117] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about e7313af6: FL 98 TL 10263186 HSR 2762600672 J 838876573 LSR 3367219470 DLSR 50549698 [1557513249.259131] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513249.259142] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Confirmed peer address as 72.66.83.8:5096 t[1557513249.562239] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Confirmed peer address as 72.66.83.8:5095 SWITCH NETWORKS [1557513253.843937] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513253.844019] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 1364757765 PC 1063357287 OC 2194907961 NTP TS 2472417766/ 1884000851=2472417766.438653 [1557513253.844029] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 77b8933: FL 163 TL 2305413 HSR 2377964402 J 118139039 LSR 2 460485565 DLSR 3416921415 [1557513253.844040] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513254.590466] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513254.590508] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2527484095 PC 1878948628 OC 3882176963 NTP TS 213338522/6 38146641=213338522.148580 [1557513254.590517] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about b7800717: FL 93 TL 16503124 HSR 338254320 J 1043490285 LSR 801212789 DLSR 2849120558 [1557513254.590530] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513258.679347] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SRTCP output wanted, but no crypto suite was negotiated [1557513258.679385] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513258.679411] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 1747525463 PC 239751946 OC 2859161225 NTP TS 3835334254/3 577221347=3835334254.832887 [1557513258.679418] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about e03bd142: FL 40 TL 11515285 HSR 3870936491 J 4150119236 LSR 2903910302 DLSR 700479064 [1557513258.679427] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513259.940740] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513259.940770] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513259.940793] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2745657868 PC 580822102 OC 4072320279 NTP TS 252333929/20 93935903=252333929.487532 [1557513259.940801] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 476ec510: FL 61 TL 5399464 HSR 3682903462 J 1600791504 LSR 497094189 DLSR 826539320 [1557513259.940814] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513263.532695] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513263.532722] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 301750475 PC 2434799802 OC 863475295 NTP TS 3625503376/30 14095725=3625503376.701774 [1557513263.532729] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 228989b7: FL 148 TL 10638176 HSR 2137788972 J 2678688015 LS R 2118438561 DLSR 1024262683 [1557513263.532738] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.051305] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513264.051352] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1190725747 PC 3072491822 OC 537479863 NTP TS 817717026/42 82321058=817717026.997056 [1557513264.051359] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 920db8ff: FL 98 TL 6598240 HSR 2045142812 J 3171281598 LSR 2224002690 DLSR 2514604823 [1557513264.051369] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513264.058891] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513264.058920] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 5c8f2711: FL 24 TL 1700303 HSR 2993289918 J 618517078 LSR 1 835926259 DLSR 3106454771 [1557513264.058891] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513264.058920] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 5c8f2711: FL 24 TL 1700303 HSR 2993289918 J 618517078 LSR 1 835926259 DLSR 3106454771 [1557513264.058934] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513264.266612] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'offer' from 127.0.0.1:60736 [1557513264.266654] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'offer' from 127.0.0.1:60736: { "sdp": "v=0 o=- 3766502030 3766502031 IN IP4 192.168.1.200 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5097 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 10.255.81.66 b=TIAS:64000 a=rtcp:5098 IN IP4 10.255.81.66 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2009301833 cname:32ababda5d2c847b a=setup:actpass a=fingerprint:SHA-256 4F:69:B6:89:51:98:6C:A4:63:CF:F7:CD:22:D2:BE:FD:6E:3F:F4:2A:8B:39:AD:E4:C0:8D:AE:BD:A3:70:12:49 ", "ICE": "remove", "DTLS": "passive", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9 799-0557b76cc340", "received-from": [ "IP4", "97.47.70.223" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d 00", "command": "offer" } [1557513264.266814] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Default sink codec is speex/16000 [1557513264.266826] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/16000 [1557513264.266831] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/8000 [1557513264.266835] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/32000 [1557513264.266838] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec GSM/8000 [1557513264.266842] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMU/8000 [1557513264.266846] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513264.266850] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec G722/8000 [1557513264.267614] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268089] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268146] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268220] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 10.255.81.66:5097 [1557513264.268627] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 10.255.81.66:5098 [1557513264.269072] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: offer time = 0.002384 sec [1557513264.269084] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'offer' from 127.0.0.1:60736 [1557513264.269098] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'offer' to 127.0.0.1:60736: { "sdp": "v=0 o=- 3766502030 3766502031 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20064 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2009301833 cname:32ababda5d2c847b a=rtpmap:104 iLBC/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20065 a=setup:actpass a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513264.355109] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SRTCP output wanted, but no crypto suite was negotiated [1557513264.355140] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513264.355150] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 511a36de: FL 144 TL 3729734 HSR 2556650506 J 4034165594 LSR 900570716 DLSR 3118967760 [1557513264.355166] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513264.355186] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513264.355193] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about aea81208: FL 180 TL 10378624 HSR 1586287909 J 2246375086 LS R 3231023203 DLSR 4073439206 [1557513264.355201] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.355210] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Switching local interface to 209.182.216.71:20033 [1557513264.358206] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513264.358229] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about b59d47bc: FL 242 TL 13886413 HSR 3966401064 J 2679493971 LS R 3040876731 DLSR 2668321392 [1557513264.358239] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.358262] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: SRTP output wanted, but no crypto suite was negotiated [1557513264.425636] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'answer' from 127.0.0.1:45431 [1557513264.425679] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'answer' from 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502031 3766502033 IN IP4 100.81.62.76 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5079 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 100.81.62.76 b=TIAS:64000 a=rtcp:5080 IN IP4 100.81.62.76 a=sendrecv a=rtpmap:104 iLBC/8000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2006816651 cname:1c0965113e9b1046 a=setup:active a=fingerprint:SHA-256 AD:2D:0F:A6:11:DD:75:CC:3C:50:3A:9A:F0:29:20:8D:95:BC:3E:2A:A0:34:F6:67:14:A7:87:EB:8D:66:77:B1 ", "ICE": "remove", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "a nswer" } [1557513264.426806] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: answer time = 0.001099 sec [1557513264.426820] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'answer' from 127.0.0.1:45431 [1557513264.426831] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'answer' to 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502031 3766502033 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20052 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2006816651 cname:1c0965113e9b1046 a=rtpmap:104 iLBC/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20053 a=setup:passive a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513264.700442] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513264.701235] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513264.892393] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513264.893153] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS handshake successful [1557513264.893193] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS-SRTP successfully negotiated [1557513264.893202] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, incoming: [1557513264.893211] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key dob4sywpHJJ8qNgFzemlzQ== salt y7/1RgzeAtPkG+GfMLI= [1557513264.893216] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, outgoing: [1557513264.893222] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key ontM1nC56XojOkmzb8520g== salt Hom6TrHovUBUaF1g6Iw= [1557513264.893228] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513269.223923] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Confirmed peer address as 97.47.70.223:4667 [1557513269.337491] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513269.337543] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 402044265 PC 1976845075 OC 2962195809 NTP TS 1329245270/2 977782988=1329245270.693319 [1557513269.337552] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about c1e9445b: FL 196 TL 15740765 HSR 1231260891 J 3373413243 LS R 2989889476 DLSR 1721296716 [1557513269.337562] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513269.337571] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Confirmed peer address as 97.47.70.223:4640 [1557513269.355712] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513269.355744] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 46888194 PC 1178262997 OC 1943197043 NTP TS 748304878/374 [1557513269.355749] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 4d78530a: FL 59 TL 15587360 HSR 641166863 J 455122196 LSR 2 981586141 DLSR 1538485973 [1557513269.355755] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513269.355763] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Confirmed peer address as 174.204.14.29:10496 [1557513269.355768] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Switching local interface to 209.182.216.71:20065 [1557513269.376119] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513269.376149] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 60b442ee: FL 185 TL 2381849 HSR 1868471658 J 3454737373 LSR 3540730944 DLSR 753951698 [1557513269.376164] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513269.967467] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: Confirmed peer address as 174.204.14.29:10523 s[1557513274.064040] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513274.064063] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513274.064084] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2043506601 PC 348629863 OC 1509649344 NTP TS 4002640369/1 330623176=4002640369.309810 [1557513274.064090] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 1b7ce7ec: FL 17 TL 10773844 HSR 2006802227 J 3496394774 LSR 1855863287 DLSR 207902258 [1557513274.064105] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 1b7ce7ec) - deleting SSRC 9fda8 18d [1557513274.064114] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding w[1557513274.337285] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513274.337341] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 3588538011 PC 4113250253 OC 3762757735 NTP TS 3675530209/ 2088363704=3675530209.486235 [1557513274.337352] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 1904c0d9: FL 190 TL 1277330 HSR 3700606795 J 2076634040 LSR 2826708796 DLSR 3084916091 [1557513274.337367] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 1904c0d9) - deleting SSRC d4b12 78d [1557513274.337378] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding itch complete[1557513278.796840] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513278.796890] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1179546147 PC 253436483 OC 401514196 NTP TS 591585900/307 4094117=591585900.715743 [1557513278.796900] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about f25fab39: FL 196 TL 1861219 HSR 1237831119 J 2726348211 LSR 3841137799 DLSR 843496811 [1557513278.796913] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add f25fab39) - deleting SSRC bb949 38e [1557513278.796921] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513279.310594] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SRTCP output wanted, but no crypto suite was negotiated [1557513279.310626] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513279.310646] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 624630617 PC 1167418798 OC 2870042862 NTP TS 2807429790/9 64181525=2807429790.224491 [1557513279.310652] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about bb84140b: FL 193 TL 5406510 HSR 1470777656 J 207786249 LSR 3684806947 DLSR 307709608 [1557513279.310669] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add bb84140b) - deleting SSRC 3813d 27e [1557513279.310679] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513279.678189] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: SRTP output wanted, but no crypto suite was negotiated [1557513283.537471] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513283.537514] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1085702245 PC 129819128 OC 1350967259 NTP TS 967851883/21 63508869=967851883.503731 [1557513283.537521] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 169e1507: FL 46 TL 3777017 HSR 3092777669 J 3495001522 LSR 3630093505 DLSR 120738828 [1557513283.537537] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 169e1507) - deleting SSRC 468fa 6ae [1557513283.537546] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513284.300047] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513284.300094] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 3660274611 PC 3339904092 OC 976632286 NTP TS 3250964209/2 354170892=3250964209.548123 [1557513284.300103] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about a9cab1ac: FL 9 TL 7152183 HSR 1071025251 J 612018816 LSR 12 43749409 DLSR 721434160 [1557513284.300119] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add a9cab1ac) - deleting SSRC e7313 af6 [1557513284.300130] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding bye [1557513286.941239] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'delete' from 127.0.0.1:49479 [1557513286.941278] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'delete' from 127.0.0.1:49479: { "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "delete" } [1557513286.941434] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Scheduling deletion of call branch 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' (via-branch '') in 3 0 seconds [1557513286.941444] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: delete time = 0.000152 sec [1557513286.941474] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'delete' from 127.0.0.1:49479 [1557513286.941618] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'delete' to 127.0.0.1:49479: { "created": 1557513231, "created_us": 973175, "last signal": 1557513264, "SSRC": { "3046983612": { }, "2450372863": { }, "1622426350": { }, "2930250248": { }, "419741913": { }, "579439031": { }, "136 0672478": { }, "3145995275": { }, "125536563": { }, "2848633260": { }, "3762016578": { }, "379458823": { }, "1552885521": { }, "3078620951": { }, "12997 31210": { }, "461170668": { }, "2009301833": { }, "3253290075": { }, "1198441744": { }, "4066356025": { }, "2006816651": { } }, "tags": { "c71ac9de-9bf5- 4f39-9fda-6bd3aaf97d00": { "tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "created": 1557513231, "in dialogue with": "2684eff8-3602-41ed-aca2-9f7938362462", "me dias": [ { "index": 1, "type": "audio", "protocol": "UDP/TLS/RTP/SAVP", "streams": [ { "local port": 20064, "endpoint": { "family": "IPv4", "address": "174.204. 14.29", "port": 10523 }, "advertised endpoint": { "family": "IPv4", "address": "100.81.62.76", "port": 5079 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513286, "flags": [ "RTP", "filled", "confirmed", "kernelized", "no kernel support", "DTLS fingerprint verified" ], "SSRC": 2006816651, "stats": { " packets": 595, "bytes": 40090, "errors": 0 } }, { "local port": 20065, "endpoint": { "family": "IPv4", "address": "174.204.14.29", "port": 10496 }, "advertised endpoint": { "family": "IPv4", "address": "100.81.62.76", "port": 5080 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513284, "flags": [ "RTCP ", "filled", "confirmed", "kernelized", "no kernel support" ], "SSRC": 2006816651, "stats": { "packets": 15, "bytes": 1306, "errors": 0 } } ], "flags": [ "initi alized", "send", "recv", "DTLS-SRTP", "DTLS role passive", "ICE controlling" ] } ] }, "2684eff8-3602-41ed-aca2-9f7938362462": { "tag": "2684eff8-3602-41ed-aca2- 9f7938362462", "created": 1557513231, "in dialogue with": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "medias": [ { "index": 1, "type": "audio", "protocol": "UDP/TL S/RTP/SAVP", "streams": [ { "local port": 20052, "endpoint": { "family": "IPv4", "address": "97.47.70.223", "port": 4667 }, "advertised endpoint": { "family": " IPv4", "address": "10.255.81.66", "port": 5097 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513284, "flags": [ "RTP", "filled", "confirmed", "kernelized", "no kernel support", "DTLS fingerprint verified" ], "SSRC": 2009301833, "stats": { "packets": 55, "bytes": 3410, "errors": 0 } }, { "local port": 20053, "endpoint": { "family": "IPv4", "address": "97.47.70.223", "port": 4640 }, "advertised endpoint": { "family": "IPv4", "address": "10.255.81.66", "port": 5098 }, "last packet": 1557513283, "flags": [ "RTCP", "filled", "confirmed", "kernelized", "no kernel support" ], "SSRC": 2009301833, "stats": { "packets": 10, "bytes": 936, "errors": 0 } } ], "flags": [ "initialized", "send", "recv", "DTLS-SRTP", "DTLS role passive" ] } ] } }, "totals": { "RTP": { "packets": 650, "bytes": 43500, "errors": 0 }, "RTCP": { "packets": 25, "bytes": 2242, "errors": 0 } }, "result": "ok" } [1557513287.087244] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513287.087244] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513287.087282] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1725021880 PC 1560319321 OC 3364521432 NTP TS 1121433750/3398500622=1121433750.791275 [1557513287.087290] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 9ba3be3b: FL 18 TL 4080170 HSR 3714308681 J 2085763389 LSR 1453166610 DLSR 3867570679 [1557513287.087303] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 9ba3be3b) - deleting SSRC 77b8933 [1557513287.087311] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513287.087324] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513287.087328] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 477b6daa: FL 136 TL 7495912 HSR 3723043622 J 76729872 LSR 791112841 DLSR 2264873880 [1557513287.087332] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 477b6daa) - deleting SSRC b7800717 [1557513287.087338] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513287.165301] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513287.165344] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 4203649664 PC 1900914033 OC 1842228915 NTP TS 2577169143/1958584749=2577169143.456018 [1557513287.165355] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 643dc91f: FL 113 TL 8548413 HSR 913668020 J 2977588542 LSR 2353716277 DLSR 1468396047 [1557513287.165370] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 643dc91f) - deleting SSRC e03bd142 [1557513287.165382] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513287.165396] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513287.165403] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 7d2457ab: FL 45 TL 2024736 HSR 174032952 J 3137856066 LSR 2564654212 DLSR 844577710 [1557513287.165410] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 7d2457ab) - deleting SSRC 476ec510 [1557513287.165418] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513316.000248] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Call branch 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' (via-branch '') deleted, no more branches remaining [1557513316.000290] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Final packet stats: [1557513316.000297] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --- Tag '2684eff8-3602-41ed-aca2-9f7938362462', created 1:25 ago for branch '', in dialogue with 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' [1557513316.000308] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000318] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20052 <> 97.47.70.223:4667 , SSRC 77c38349, 55 p, 3410 b, 0 e, 32 ts [1557513316.000325] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20053 <> 97.47.70.223:4640 (RTCP), SSRC 77c38349, 12 p, 1120 b, 0 e, 29 ts [1557513316.000330] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --- Tag 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00', created 1:25 ago for branch '', in dialogue with '2684eff8-3602-41ed-aca2-9f7938362462' [1557513316.000339] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000347] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20064 <> 174.204.14.29:10523, SSRC 779d978b, 595 p, 40090 b, 0 e, 30 ts [1557513316.000355] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20065 <> 174.204.14.29:10496 (RTCP), SSRC 779d978b, 17 p, 1490 b, 0 e, 29 ts [1557513316.000521] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet

2 1

DMQ re-ordering concern
by Julien Chavanton 11 May '19

11 May '19

Yesterday at Kamailio World there was an interesting question, raising a concern about replication of user location and DMQ in general, we should track it in the mailing list and use it as a feature request and I can share my initial thoughts on the topic. The concern was about DMQ reordering transactions. With SIP registration this seems like a minor concern, high responsiveness at the cost of potential small inconsistency seems like a good option. Considering that, the state will automatically correct itself and that this can be controlled using the expiration timer. This will become even more insignificant when the replicas can only be used as primary server failover. (it becomes a very small concern only when the primary server fails) With other type of data re-ordering may result in more problematic use cases. My impression is that DMQ should try to be good only with volatile data, data that will expire anyway, like caching with expiration, registration, Dialogs, etc. I think it may be worth looking at adding support for ordering/queuing in DMQ. Performance and simplicity could be maintained by doing mainly transactions in parallel with a configurable re-ordering best effort to minimize the impact of the problem. For example : 1- trying to re-order for a configurable amount of time. (32 seconds to match the RFC for SIP re-transmissions) 2- adding tractability for failed transactions. 3- the possibility to trigger a re-sync. Another option is to look at preserving strict ordering but I can imagine this could add more problems in some cases ...

2 4

Question regarding NAPTR and RURI
by Schneuwly Mathias RUAG 10 May '19

10 May '19

Hi I've a question about NAPTR lockup and rewriting the RURI. Following a little example: - Alice at location A tries to call Bob at location B - location A uses the RURI naptr.locationB.com to reach location B - this NAPTR resolves to hostb.locationB.com and host B is listening on port 5070 hostb has the following alias defined: - udp:hostb.locationB.com:5070 The problem now is, that the incoming RURI is "sip:bob@naptr.locationB.com" and the check to myself is not True. Is there a possibility to rewrite the RURI to the result of the NAPTR query? If I would add a second alias like "udp:naptr.locationB.com:5070" a check to myself would still fail as the RURI does not contain a port. So I would have to add an alias like "udp:naptr.locationB.com" without port, then it would work.? Is this the way to go? Because the documentation says it is important to include the port, otherwise the loose_route() will not work as expected... Thanks for your replies! Mathias

1 0

Help identifying mem leaks
by Duarte Rocha 10 May '19

10 May '19

Greetings, Recently one of my Kamailio's started started answering with a lot of 500 replies to requests because of insufficient private memory. The service Kamailio had been running for 6 months now. In order to solve this i increased the private memory settings and restarted the Kamailio. Before that, when searching a bit about the issue i found an old thread where Daniel suggested using " kamcmd corex.pkg_summary idx 1" to check for pkg leaks in the sip worker process. I ran this before doing the restart in order to identify leaks and found the summary in the logs. However, i am having a hard time interpreting this summary and being able to know if i have or not a mem leak. What should i be looking for? This is the beggining of the summary : qm_status: (0x7ff3c8cef010): qm_status: heap size= 8388608 qm_status: used= 928576, used+overhead=1464080, free=6924528 qm_status: max used (+overhead)= 1488472 qm_status: dumping all alloc'ed. fragments: qm_status: 0. N address=0x7ff3c8d286d8 frag=0x7ff3c8d286a0 size=1024 used=1 qm_status: alloc'd from core: core/str_hash.h: str_hash_alloc(59) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 1. N address=0x7ff3c8d28b40 frag=0x7ff3c8d28b08 size=256 used=1 qm_status: alloc'd from core: core/str_hash.h: str_hash_alloc(59) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 2. N address=0x7ff3c8d28ca8 frag=0x7ff3c8d28c70 size=184 used=1 qm_status: alloc'd from core: core/counters.c: cnt_hash_add(332) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Any help? Best Regards, Duarte Rocha

1 0

How to configure Kamailio 5.1 with Asterisk pjsip
by przeqpiciel 10 May '19

10 May '19

Hi, There is any working examples how to configure dispatcher module of Kamailio? I have working registrations but working in half dialing function.Signalling are come to asterisk and rtp start sending to my IP but i cant hear anything. Asterisk are doing registrar job - i dont know if it is good idea, maybe not.

5 11

RTP multiplexing, batching and compression (KamailioWorld followup)
by Stefan Sayer 09 May '19

09 May '19

Hi, as a followup to the 5 min 5 slides talk on Tuesday at the Kamailio world conference, I wanted to send links to that implementation, as I did get some interest about this afterwards. This transparent RTP mux gateway implementation in SEMS is a multiplexing, batching and compression mechanism, which can save quite some bandwidth on low bitrate codecs - can be useful if you need to work with very low bitrate links, e.g. on satellite (I tested with 10 G729 calls instead of 5 on a 128kbit sat link). It's a bit like ROHC (RFC3095) but on application level, only for RTP, and by using configuration for IP addresses instead of negotiation. Or as osmux, but without separate negotiation (but in-band), and for all codecs, and SIP. Or iax2 trunking mode, but with normal SIP/RTP. https://github.com/sems-server/sems/blob/sayer/rtp-mux/doc/rtp_mux/rtp_mux_… https://github.com/sems-server/sems/blob/sayer/rtp-mux/doc/rtp_mux/rtp_mux.… For now it's in the sayer/rtp-mux branch, but it's planned to merge it to master, possibly with simplification of the configuration. On another note, many thanks to the organizers of the Kamailio World conference! It's both really nice to meet, and very useful, and I think as well individually as well as for the community as a whole. Best Regards, Stefan

1 0

CRM / Windows integration
by Mark Boyce 09 May '19

09 May '19

Hi All Does anyone have any pointers for integrating Kamailio with CRMs (Dynamics / Salesforce / Zoho) or native Windows? We’re looking for basic click to dial / screen pop. Thanks Mark -- Mark Boyce Dark Origins Ltd

4 5

RE-invite in case of SIP is Active
by Gaurav Bmotra 09 May '19

09 May '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 How can i use Re-invite in kamailio if SIP is active so that call can reconnect in case of network loss plz help -- *Regards:* Gaurav Kumar

2 2

Call reconnect in case of loss network
by Gaurav Bmotra 09 May '19

09 May '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 when i call from one to another extension , in case of loss network it disconnect the call or stuck there , how can i make it as re-connection when network loss , and it will try to reconnect for some seconds like 30 sec "Example for callingn skype it reconnectcall in case of lossnetwork " plz help -- *Regards:* Gaurav Kumar

1 0

Sending in-dialog requests from custom module's C code
by Ivan Ribakov 09 May '19

09 May '19

I’m working on a custom Kamailio module where I need to send new-dialog and in-dialog requests on timer (hence I’m being forced to generate new messages from C code). So far I’ve been using modules/tm/tm_load.h defined interface to generate messages and handle callbacks. New dialog messages are sent and processed normally. To do that I’m: 1. Calling set_uac_request() to define request parameters 2. Calling tmb.t_request_outside() to send it outside any existing dialog I was able to send in-dialog requests (or so I thought) in a similar fashion, but I soon realised that responses to those requests were dropped because they couldn’t be matched against any known existing transaction. I’m attaching log messages that I believe support this theory and I’ve also observed UDP retransmissions. tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 50576 label 0 branch 0 tm [t_lookup.c:990]: t_reply_matching(): no matching transaction exists tm [t_lookup.c:993]: t_reply_matching(): failure to match a transaction tm [t_lookup.c:1088]: t_check_msg(): msg (0x5abcb50) id=1 global id=1 T end=(nil) tm [t_reply.c:2195]: reply_received(): transaction not found - (branch -1) The way I’m currently generating in-dialog requests is very similar to what tmb.t_request_outside() does, the main difference being that I do the dialog setup manually, based on the call-ID, cseq and from/to tags (I’m tracking transaction identifiers separately) and then pass resulting uac_req_t to tmb.t_request_within() - https://gist.github.com/IvanRibakov/3302cb286b1f4b786d109b406f2435a2 <https://gist.github.com/IvanRibakov/3302cb286b1f4b786d109b406f2435a2> Now, the question part - does anyone know what I’m doing wrong/missing? As I mentioned, when looking at the generated message bodies, they look ok to me (left - initial request that started the dialog, right - first in-dialog request), so I’m guessing I’m missing some Kamailio internal steps needed to register new transaction. SIP flow (up to the point when first UDP retransmission happens) I apologise in advance for the bulky question and will be extremely thankful for any guidance. Regards, Ivan

2 2

GDB debugging Kamailio 5.2.2 IMS authorization failed by the SCSCF
by Woscek, Martin W. 08 May '19

08 May '19

Hi, Looking for some advice running the debugger and setting breakpoints in Kamailio modules such as the SCSCF. Example: We were trying to watch the authenticate() function: 714 int authenticate(struct sip_msg* msg, char* _realm, char* str2, int is_proxy_auth) { Our Kamailio 5.2.2 code was compiled and linked from source with the -g option already enabled by default. We also dropped our Workers to 1 from 16 in the scscf.xml file for the SCSCF to limit the thread count and get the following list of threads below: We tried the first cdp threads: 8462, 8463, and 8464 and while we could set a break point at line 714 for the authenticate() function, it didn't actually break there at all. Question: What pid will we use to catch the authenticate() called during an initial registration from a boghe client? Anything special we may be missing, gdb related or Kamailio settings? kamcmd -s /run/kamailio_scscf/kamailio_ctl kamcmd 1.5 Copyright 2006 iptelorg GmbH This is free software with ABSOLUTELY NO WARRANTY. For details type `warranty'. kamcmd> ps 8456 main process - attendant 8457 udp receiver child=0 sock=10.0.20.1:6060 8458 slow timer 8459 timer 8460 secondary timer 8461 ctl handler 8462 cdp worker child=0 8463 cdp receiver peer unknown 8464 cdp_receiver_peer=hss522.irisims.org 8465 cdp_acceptor 8466 cdp_timer 8470 notification_worker 8471 notification_worker 8472 notification_worker 8473 notification_worker 8477 JSONRPCS FIFO 8478 JSONRPCS DATAGRAM Any advice appreciated. Thanks, Martin Woscek

2 1

http_client_query module?
by Andrew Chen 08 May '19

08 May '19

Hey guys, Need help here. What module besides tls.so I need to load in order to get http_client_query to work? The module doc says tls.so: #!ifdef WITH_TLS loadmodule "tls.so" #!endif but I keep getting this when doing config check: root@ashmaintpsg51:/etc/kamailio# kamailio -c kamailio.cfg 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 322, column 71: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 323, column 45: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 324, column 42: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 325, column 48: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 326, column 44: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 327, column 71: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 328, column 67: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 329, column 41: Can't set module parameter 0(12971) ERROR: <core> [core/cfg.y:3309]: yyparse(): cfg. parser: failed to find command http_client_query (params 2) 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 859, column 105: unknown command, missing loadmodule? and I have the package installed on my ubuntu 18.04: ii kamailio-*tls*-modules:amd64 5.1.2-1ubuntu2 amd64 TLS support for the Kamailio SIP server (authentication, transport) thoughts? -- Andy Chen Sr. Telephony Lead Engineer 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 3

Udp errors on High Availability Setup
by Duarte Rocha 08 May '19

08 May '19

Greetings, I have two Machines with Kamailio in a High Availability (HA) configuration. Both machines are active and processing calls and both have one HA IP that can jump to the other machine in case of failure. I also have probing for NAPs configured with each IP as the socket. So, in each machine i have this configuration : listen=udp:HA_Address1 listen=udp:HA_Address2 NAP 1 being probed from socket HA_Address1 NAP 1 being probed from socket HA_Address2 I also have /proc/sys/net/ipv4/ip_nonlocal_bind with value 1. Everything works correctly with this configuration. However when the probing is sent for the NAP' my logs get filled with the same error messages. udp_send(): sendto(sock, buf: 0x7f2d4c91fa18, len: 340, 0, dst: (xxx.xxx.xxx.xxx:xxx), tolen: 16) - err: Invalid argument (22) CRITICAL: <core> [core/udp_server.c:602]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed The error happens because it is trying to probe a NAP using a socket that isn't active at the moment in the machine. Is there a way for this error not to be logged or be disabled without reducing Log Level to suppress CRITICAL errors? I think this is a standard High Availability configuration. Is there some setting that can disable this kind of errors? Best Regards, Duarte Rocha

3 5

HELP ME
by khadime gaye 07 May '19

07 May '19

Good morning, How I can resolve this problem ? << ERROR: Could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' >>

2 1

implementing custom authentication method
by Walter.Martín Villalba 03 May '19

03 May '19

Dear list, I'm implementing a custom method for user authentication and would like to kindly ask for some guidance regarding the server side changes that may be needed in order to keep things running smoothly. The idea is to replace the standard user/pass-based digest authentication with a custom one based on Open ID Connect (OIDC) access tokens. I already have some pieces working (receiving access token in REGISTER message, validating the token by calling custom python routine, etc.), but I'm missing the bits needed to keep track of users who have successfully authenticated. While looking at sipcomm.cfg I found calls to *www_authenticate* (defined in modules/auth_db/authorize.c), which in turn calls to *mark_authorized_cred*, *check_auth_hr* (or *auth_check_hdr_md5*), and *generate_avps*,. Some of these functions are indeed changing some values in the DB and/or in memory, and this is where I would like some guidance. Is there any documentation I could look at which hopefully explains what DB values and/or data structures in memory need to be updated in order to keep track of successfully authenticated users? I believe with this last piece I'd be able to wrap up my custom authentication implementation. Thanks in advance! Regards, Walter Martín.

1 0

Kamailio bypassing RTP media
by Ali Taher 03 May '19

03 May '19

Hi everyone, I'm new to kamailio. I plan to setup a sip server cluster, Does someone can give me some suggestions if I can use kamailio as front server, which handle sip message and bypass rtp media message to backendserver, this mean the kamailio handle only sip message, the rtp packets are sent directed to backend server. The sip sequences is 1.softswitch send invite to kamailio 2.kamailio send invite to backend server. 3.backend server return 200 ok to kamailio 4.kamailio return 200 ok to softswitch, the ip and port in sip is from the backend server's msg, this make the softswitch send rtp direct to backend server. 5.rtp packets are transfered between softswitch and backend server. .... 6.backend server send bye to kamailio 7.kamailio send bye to softswitch. Any help is appreciated. Regards, Ali Taher

4 5

forcing socket doesn't work for ACK
by David Villasmil 03 May '19

03 May '19

Hello guys, I have public and private IPs, and i need to force the sending socket replying to internal out externals endpoint, so i'm using this: function to check the destination (i tried using $sndto(ip) but it always returns NULL): route[CHECK_DEST_NET] { if ( $du=~"sip:172\..*" ) { setflag(FLAG_TO_PRIVATE); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PRIVATE -> [$du]\n" ); } else { setflag(FLAG_TO_PUBLIC); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PUBLIC -> [$du]\n" ); } return; } on my relay: route(CHECK_DEST_NET); if (isflagset(FLAG_TO_PRIVATE)) { xlog("L_ERR", "[RELAY] forcing socket to PRIVATE NET\n" ); force_send_socket(LISTEN_INSIDE_IF:LISTEN_INSIDE_PORT); } else { xlog("L_ERR", "[RELAY] forcing socket to PUBLIC NET\n" ); force_send_socket(LISTEN_OUTSIDE_IF:LISTEN_OUTSIDE_PORT); } This works nicely, except for ACKs and potentially other packets sent statelessly? Anyone knows how to fix this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

6 26

run some extra things before relying message
by Mojtaba 02 May '19

02 May '19

Hello, I have encountered with problem in Kamailio. At first glance it looks simple. Considering the Data Lump feature in Kamailio, All changes in SIP body, does not apply quickly. So if we are using RTPEngine or RTPProxy, How we can get body message (Actually IP and port from SDP after rtpengine_manage function) before relaying message? Suppose i want get some information and run some extra things before relying message? I just added my extra things in t_on_reply block, but it's strange why it doesn't work before relaying! -Mojtaba Esfandiari.S

2 2

Problem with topoh
by Phillman25 Kyriacou 02 May '19

02 May '19

Dear List I have an issue with topoh module. Specifically for in-dialog requests, if the encoded uri is altered i.e. initial encoded contact uri is Contact: <sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W> and UAS sends an in-dialog INVITE like this adding tgrp and user=phone: INVITE sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W;tgrp=1069912311;user=phone Kamailio cant seem to match/decode this and responds "404 Not Found" I have also set: modparam("topoh", "uri_prefix_checks", 0) As specified here: https://www.kamailio.org/docs/modules/4.4.x/modules/topoh.html#idp37582868 But still doesn't work. Is there a bug with this module? # ==================================================== # ================ TOPOLOGY HIDING =================== # ==================================================== modparam("topoh", "mask_key", "secret") modparam("topoh", "mask_ip", "X.X.X.X") modparam("topoh", "mask_callid", 0) modparam("topoh", "uparam_name", "line") modparam("topoh", "uparam_prefix", "sr-") modparam("topoh", "vparam_name", "branch") modparam("topoh", "vparam_prefix", "z9hG4bKsr-") modparam("topoh", "uri_prefix_checks", 0) Thanks in advance for your support Phillip

2 4

Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 02 May '19

02 May '19

Hello, I was trying out the auth_ephemeral module inside the xhttp event_route in Python KEMI, and discovered a possible bug. https://pastebin.com/4Ghumfis The following will result in the stack smash error. Whenever I leave out the unixtimestamp the module will complain about not being able to convert to an int, which make sense, but doesn't lead to a stack smash error and a complete crash of Kamailio. KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). According to the documentation this module can only be used in REQUEST_ROUTE, but I found a presentation online which also used this module and that particular function in event_route, which is what I need for WebSocket authentication. Hope this helps. Please let me know if you need more information. Regards, Grant

2 1

Kamailio 5.2.2 IMS authorization failed by the SCSCF
by Woscek, Martin W. 02 May '19

02 May '19

Hi, We are getting a SIP 401 Unauthorized -Challenging the UE error originating by the SCSCF. It looks to be related to the nonce/cnonce. Is there something we have misconfigured in our SCSCF module? Log output: May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18489]: DEBUG: ims_auth [authorize.c:786]: authenticate(): Nonce or response missing: nonce len [-108158976], response16 len[1873606560] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:168]: get_nonce_response(): Calling find_credentials with realm [ims522.irisims.org] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:195]: get_nonce_response(): Found nonce response May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: INFO: ims_auth [authorize.c:827]: authenticate(): uri=sip:irisims.org nonce=9863c63855c8dd888c5c18e79fc61156 response=ce7ce73872b92aff2defa76dd1aff1ac qop=auth-int nc=00000001 cnonce=4fe4daacd69fa3d609c8b396d331eae6 hbody=d41d8cd98f00b204e9800998ecf8427e May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [rfc2617.c:135]: calc_response(): calc_response(_ha1=7a1b479a0513914e71bcd6d005b3e17d, _nonce=9863c63855c8dd888c5c18e79fc61156, _nc=00000001,_cnonce=4fe4daacd69fa3d609c8b396d331eae6, _qop=auth-int, _auth_int=1,_method=REGISTER,_uri=sip:irisims.org,_hentity=d41d8cd98f00b204e9800998ecf8427e) Thanks, Martin

2 1

New voisonics.com - Kamailio related products and services
by David Cunningham 02 May '19

02 May '19

Hello, Usually it's our client's VoIP technology being updated. Fortunately we found a little spare time to completely revamp the Voisonics website as well! https://voisonics.com/ Voisonics provides products and services in the commercial VoIP and hosted PBX sector; in particular relating to Kamailio, Asterisk, and Enswitch. The business has been specialising in this area for almost 10 years. Please feel free to contact me off-list with any questions. Kind regards, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

1 0

Support for JWT token authentication?
by David Dean 01 May '19

01 May '19

I came across this thread from Jan 2015 discussing the addition of JWT token authentication to Kamailio: https://github.com/kamailio/kamailio/issues/29 Was JWT authentication ever added? Or has anyone else on this list got JWT working somehow?

1 0

CSeq not incrementing - app_ruby
by Andrew White 01 May '19

01 May '19

Hi all, I’m still building with app_ruby and loving it - for the most part! I’m having an issue with one of my providers responding with a 482 on auth invite. Researching, it appears that my CSeq isn’t incrementing and this is the reason for the issue. I already have the relevant flag set in my kamailio.cfg however: modparam("dialog", "track_cseq_updates", 1) During my uac_auth, I’ve tried manually setting the diff, as well as running msg_iflag_reset (as suggested by Daniel - https://github.com/kamailio/kamailio/issues/1359 <https://github.com/kamailio/kamailio/issues/1359> - unfortunately the function isn’t exported in KEMI): if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") #KSR::COREX.msg_iflag_reset("UAC_AUTH") KSR::PV.sets("$dlg_var(cseq_diff)", "1") KSR.info("CSeq diff: #{KSR::PV.gete("$dlg_var(cseq_diff)")}") KSR::TM.t_relay() Unfortunately in both cases the actual CSeq doesn’t increment in the second INVITE: INVITE 1 (unauthenticated): > CSeq: 102 INVITE INVITE 2 (with Authorization header): > CSeq: 102 INVITE I’m unsure if I’ve missed something silly around the track_cseq_updates flag, or if app_ruby somehow isn’t interacting with the dialog module when running uac_auth to trigger the increment? Thanks! Kind regards, Andrew

3 6

listening at link local IPv6 address
by Juha Heinanen 01 May '19

01 May '19

My K host is behind a mobile WiFi hotspot and I tried to make K listen at the IPv6 address fe80::6e29:95ff:fe7d:37e6/64 that the host has received from the hotspot. Using that address on listen line gives error: Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:395]: udp_init(): bind(8, 0x7ff97dfb3fe4, 28) on FE80:0:0:0:6E29:95FF:FE7D:37E6: Invalid argument Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:401]: udp_init(): might be caused by using a link local address, try site local or global I can ping that address like this: $ ping6 fe80::6e29:95ff:fe7d:37e6%wlp1s0 PING fe80::6e29:95ff:fe7d:37e6%wlp1s0(fe80::6e29:95ff:fe7d:37e6%wlp1s0) 56 data bytes 64 bytes from fe80::6e29:95ff:fe7d:37e6%wlp1s0: icmp_seq=1 ttl=64 time=0.030 ms I then tried to add %wlp1s0 suffix to the listen address, but got error: Apr 21 09:10:35 salmon sip-proxy[7896]: 0(7948) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 408, column 34: ip address, interface name or hostname expected Did I do something wrong? If not, why is it that K cannot listen at a link local address? It is something that Linux networking stack does not support? -- Juha

6 33

Pipelimit with clean_unused
by Cinthia Leung 30 Apr '19

30 Apr '19

Hello, What is your recommendation on using clean_unused with db defined pipes? >>pl_check("systemwide"); If I leave the system idle, they are eventually "cleaned" as expected. In our use case, that means failed calls unfortunately. I am trying to see if I can mix both db/static pipes with dynamic/run-time pipes. Thanks Cindy

1 1

Next hop of reply, in onreply_route
by Alex Balashov 30 Apr '19

30 Apr '19

I am fairly sure it’s been asked a number of times, but is there a way to know the destination, and more especially the transport, of the next hop of a reply message, viewable from onreply_route? For requests, we have $nh and many other PVs which in some way evidence where it’s going. And yes, I’m aware that this can be previewed from the onsend_routes*, but I need to amalgamate that knowledge with some transaction-specific data. Thanks! — Sent from mobile, with due apologies for brevity and errors.

2 2

"ERROR: reply too big" and ctl binrpc size
by Karsten Horsmann 30 Apr '19

30 Apr '19

Hi all, i use kamailio 5.0.7 in production and i noticed that i cant listen the dialogs after a few days of traffic on this server. Then i got "kamcmd dlg.list_ctx " - "ERROR: reply too big". So now my question - i want to debug my "sleeping dialogs" with dlg.list - how big should it change the values of without knowing the size of the current filled up binrpc stuff? Or is there a better option to debug dialog-module? modparam("ctl", "binrpc_max_body_size", ??) modparam("ctl", "binrpc_struct_max_body_size", ?) -- Cheers and see you in Berlin *Karsten Horsmann*

2 1

Re: [SR-Users] SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi Daniel, Thank you very much for clarifying! Cheers, Yufei On 30 Apr 2019 at 08:10, <Daniel-Constantin Mierla <miconda(a)gmail.com>> wrote: Hello, there is an errata for that RFC, but it refers only to the BYE, where it adds back the transport=tcp, but the ACK R-URI should follow the same rules, see: * https://www.rfc-editor.org/errata/eid5294 Cheers, Daniel On 28.04.19 22:54, Yufei Tao wrote: Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei _______________________________________________ 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 - May 6-8, 2019 -- www.kamailioworld.com

1 0

docker image for non x86 arch
by Sergey Safarov 30 Apr '19

30 Apr '19

Is the community involved in kamailio docker image for ARM architecture and others not x86 platforms? ARM ARM 64 IBM POWER IBM Z PowerPC 64 LE

3 3

SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei

3 2

remove_record_route - bug?
by Sergiu Pojoga 29 Apr '19

29 Apr '19

Hi ppl, After a branch route fails, I need to re-evaluate the next destination and adjust RR parameters, if need be, like for example adjust the nat=yes|priv. failure_route[MANAGE_PSTN_FAILURE] { ... remove_record_route(); record_route(); route(NATMANAGE); ... route(RELAY); } The result is that the new branch is stripped of "Record-Route: <sip:MYIP>", but the params remained. Moreover, record_route hasn't been re-added as instructed in the failure route. Request headers of the initial branch that will eventually timeout and fail: 2019/04/27 18:29:18.034992 10.22.0.1:5060 -> 10.22.0.100:5060 INVITE sip:1514XXXXXXX@10.22.0.100 SIP/2.0 Record-Route: <sip:10.22.0.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> Record-Route: <sip:65.XX.XX.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> New request branch after failure: 2019/04/27 18:29:19.538292 65.XX.XXX.1:5060 -> 65.XX.XX.2:5060 INVITE sip:1514XXXXXXX@65.XX.XX.2 SIP/2.0 ;lr=on;did=9e2.3dd2;nat=priv> kamailio -v version: kamailio 5.1.6 (x86_64/linux) 7d1964 Much obliged. --Sergiu

4 5

Vide SIP conferencing kamailio with freeswitch
by Gaurav Bmotra 28 Apr '19

28 Apr '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help -- *Regards:* Gaurav Kumar

2 1

Re: [SR-Users] [sr-dev] Vide SIP conferencing kamailio with freeswitch
by Henning Westerholt 28 Apr '19

28 Apr '19

Hello, please ask this usage related questions on our "sr-users" list. Kamailio is not doing much regarding the video calls. E.g. if your calls work for voice with kamailio, they should also work for video. All the setup needs to be done in freeswitch. I found a (outdated) howto with google: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video Maybe this helps. Henning Am 27.04.19 um 13:22 schrieb Gaurav Bmotra: HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

How to configure ipv6 in kamailio
by Santosh Kumar Dash 28 Apr '19

28 Apr '19

Hi All, I have a kamailio installed in my setup which is working properly in IPV4 configuration. I want to perform IPV6 configuration, I don't have any idea how to configure it, I have tried few things by changing *kamailio.cfg* file but did not work. I am sharing my *kamailio.cfg *file which is working fine in ipv4 Kindly help me to configure for IPV6. lets say my Ipv6 address is : 2402:3a80:d1a:f45a:fe44:2e54:ed2b:9bb2 Thanks,

3 3

Can't set module parameter
by Tmbc 26 Apr '19

26 Apr '19

Hi all Having this issue with kamailio: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 119, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 120, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 121, column 50: Can't set module parameter here is full conf file: https://dpaste.de/bP3E OS: NAME="CentOS Linux" VERSION="7 (Core)" Software version: kamailio 5.2.2 (x86_64/linux) 67f967 Thanks in advance, tmbc

2 1

Re: [SR-Users] Config to control TCP connection reuse
by R, Rajkumar (Raj) 24 Apr '19

24 Apr '19

I think elaborating on the setup may help someone to provide inputs on this. Below is the setup. Client 1 --> Kamailio proxy --> Backend SIP server Client 2 --> Kamailio proxy --> Backend SIP server Client 3 --> Kamailio proxy --> Backend SIP server After I register all three clients, see only one TCP connection is being used between "Kamailio proxy - Backend SIP server". Is there a way to achieve separate TCP connection to Backend SIP server from kamailio. thanks, raj From: R, Rajkumar (Raj) Sent: Monday, April 22, 2019 3:45 PM To: sr-users(a)lists.kamailio.org Subject: Config to control TCP connection reuse Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

2 1

DMQ_USRLOC
by Gholamreza Sabery 24 Apr '19

24 Apr '19

Hi, Recently, I started to use dmq_usrloc module. I successfully set it up. Unfortunately, the documentation does not specify the behavior of dmq_usrloc module with other Kamailio modules. For example, I see a bunch of KDMQ messages are exchanged between nodes, but how the lookup() function behaves with dmq_usrloc, or what is it's effect on a db_mode of 3! Regards

5 6

JSON error - control character 0xd
by David Dean 23 Apr '19

23 Apr '19

I'm trying to send an SDP from a SIP client to Janus using JSON over HTTP. The problem is that Janus is erroring on the SDP content with the following error: JANUS HTTP: Get SDP for echo plugin: {#012 "janus": "error",#012 "error": {#012 "code": 454,#012 "reason": "JSON error: on line 9: control character 0xd near '\"v=0'"#012 }#012} Result code 200 I tried using {s.escape.common} to escape any quotations, commas etc but it doesn't appear to replace the control characters with newlines. Any suggestions on how best to remove these control characters so that Janus will accept the SDP over JSON? Here is the relevant section of kamailio.cfg: # HTTP: Request SDP for echo test plugin sdp_get("$avp(sdp)"); $var(res) = http_connect("janus", "/janus/$var(JANUS-ID)/$var(ECHO-ID)", "application/json", "{ \"janus\" : \"message\", \"transaction\" : \"testtesttest99\", \"body\" : { \"audio\" : true }, \"jsep\" : { \"type\" : \"offer\", \"sdp\" : \"$(avp(sdp){s.escape.common})\" }}", "$avp(janus-pluginsession)"); xlog("L_INFO", "JANUS HTTP: Get SDP for echo plugin: $avp(janus-pluginsession) Result code $var(res)\n");

3 6

Direction of replies
by Duarte Rocha 23 Apr '19

23 Apr '19

Greetings, Is there a way to know in reply_route if a reply has direction downstream or upstream regarding the first INVITE? I know there are vars and methods for finding this out on requests, but i can't find anything for replies. Thanks in advance, Best Regards, Duarte Rocha

2 1

CDR start time of a canceled call
by Jan-Hendrik Dörner 23 Apr '19

23 Apr '19

Hello, I just noticed that the logged „start_time" as well as „end_time" of a canceled call is set to the „invite time“. If done this way, it is harder to calculate the „ringing time“. Is there an option to change this behaviour (i.e. to log the time of the received „bye“ instead)? (If done so, the ringing time would always be the time between start and invite …) Or is there a special reason for this behaviour? My CDR settings are as followed: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) # log unanswered / open dialogs modparam("acc", "cdr_on_failed", 1) # log also not answered calls modparam("acc", "cdr_extra", "inviteTime=$dlg_var(inviteTime)" modparam("acc", "cdr_log_enable", 1) # activate write to syslog modparam("acc", "cdr_facility", "LOG_LOCAL6") # syslog modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "time_mode", 0) #syslog timestamp modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) # needed for syslog modparam("acc", "log_missed_flag", FLT_ACCMISSED) # need for logging missed calls in syslog Regards Jan-Hendrik

1 0

Kamailio-5.2.1 - Issues in NDB_MONGODB module initialisation
by José Seabra 22 Apr '19

22 Apr '19

Hello there, I have NDB_MONGODB loaded in kamailio with following configurations: *loadmodule "ndb_mongodb"* *modparam("ndb_mongodb", "server", "name=mgs1;uri='mongodb://mongodbserver:27017/proxyregistrar'")* Kamailio breaks up on start up and prints the following messages: *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:83]: set_mod_param_regex(): 'ndb_mongodb' matches module 'ndb_mongodb'* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/sr_module.c:711]: find_param_export(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:99]: set_mod_param_regex(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: ndb_mongodb [mongodb_client.c:155]: mongodbc_add_server(): added server[mgs1]=mongodb://10.225.121.12:27017/proxyregistrar <http://10.225.121.12:27017/proxyregistrar>* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/sr_module.c:942]: init_mod_child(): idx 21 rank 8: ndb_mongodb [tcp receiver (generic) child=3]* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 23* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:739]: handle_sigs(): child process 3181 exited by a signal 11* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:742]: handle_sigs(): core was generated* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/tcp_main.c:3513]: handle_ser_child(): dead child 16, pid 3181 (shutting down?)* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:764]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa6d0e0, 23, -1, 0x0) fd_no=23 called* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [main.c:766]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: ndb_mongodb [ndb_mongodb_mod.c:128]: mod_destroy(): cleaning up * Core dump back trace output: *(gdb) bt full* *#0 0x00007f3daa134f93 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#1 0x00007f3daa156553 in mongoc_uri_get_ssl () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#2 0x00007f3daa134f88 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#3 0x00007f3daa135029 in mongoc_client_new () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#4 0x00007f3daae6d75f in mongodbc_init () at mongodb_client.c:63* * rsrv = 0x7f3db0f86728* * __FUNCTION__ = "mongodbc_init"* *#5 0x00007f3daae73c46 in child_init (rank=3) at ndb_mongodb_mod.c:113* * __FUNCTION__ = "child_init"* *#6 0x00000000005470d7 in init_mod_child (m=0x7f3db0f85e88, rank=3) at core/sr_module.c:846* * __FUNCTION__ = "init_mod_child"* *#7 0x00000000005474b6 in init_child (rank=3) at core/sr_module.c:874* * ret = 32573* *#8 0x00000000004b3840 in fork_process (child_id=3, desc=0x7ffecc20c7f0 "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>", make_sock=1) at core/pt.c:341* * pid = 0* * child_process_no = 3* * ret = -1* * new_seed1 = 1273263324* * new_seed2 = 1531871627* * sockfd = {14, 15}* * __FUNCTION__ = "fork_process"* *#9 0x0000000000424b9b in main_loop () at main.c:1631* * i = 2* * pid = 3807* * si = 0x7f3db123a988* * si_desc = "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>\000\177\000\000P\310 \314\376\177\000\000\000e|\261=\177\000\000 \313 \314\376\177\000\000\063\225{\000\000\000\000\000\027\000\000\000\000\000\000\000\000\000\000 \000\000\000\000\000\000\200\000\000\000\000\000\217e|\261=\177\000\000`~x\000\000\000\000\000\230\304\022\261=\177\000"* * nrprocs = 4* * woneinit = 1* * __FUNCTION__ = "main_loop"* *#10 0x000000000042c4e9 in main (argc=11, argv=0x7ffecc20cc08) at main.c:2675* * cfg_stream = 0x2893010* * c = -1* * r = 0* * tmp = 0x7ffecc20e6fd ""* * tmp_len = 0* * port = 0* * proto = 2496* * options = 0x7697a0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:"* * ret = -1* * seed = 173321635* * rfd = 4* * debug_save = 0* * debug_flag = 0* * dont_fork_cnt = 0* * n_lst = 0x7ffecc20cac0* * p = 0x7f3db177497d <intel_check_word+173> "H\205\300t\025\017\266P\003\017\266\312D9\371\017\204", <incomplete sequence \316>* * st = {st_dev = 19, st_ino = 21919, st_nlink = 2, st_mode = 16832, st_uid = 993, st_gid = 2, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1551179759, tv_nsec = 561093463}, st_mtim = {tv_sec = 1551179759,* *---Type <return> to continue, or q <return> to quit---* * tv_nsec = 561093463}, st_ctim = {tv_sec = 1551179759, tv_nsec = 561093463}, __unused = {0, 0, 0}}* * __FUNCTION__ = "main"* *(gdb)* NDB_MONGODB configuration is pretty simples and currently i'm not making any mongodb operations, i'm just loading the module and trying to connect against mongodb. Any thoughts on this issue? Thank you Best regards -- Cumprimentos José Seabra

2 1

rtpengine_offer() with ICE not updating o= and m= IP addresses
by David Dean 22 Apr '19

22 Apr '19

I'm using the following rtpengine_offer() to force the use of ICE relay and also replace o= and m= rtpengine_offer("replace-origin replace-session-connection ICE=force-relay RTP"); The SDP is being updated to include an ICE relay candidate, but the IP addresses in the o= and m= lines are not changing to the servers IP address (X.X.X.X). I also tried using the media-address parameter but that didn't change the behaviour. When I use rtpengine_answer() then it's not providing an ICE candidate nor changing o= or m=, which I imagine is expected. What am I missing? INVITE from the client (before rtpengine)------------------------------------------------------ INVITE sip:server.domain.com SIP/2.0Via: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rportFrom: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>Contact: <sip:username@154.20.1.8:37520;transport=tcp>Call-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Content-Length: 1506Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneSupported: 100rel --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=ptime:20a=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=rtpmap:114 AMR-WB/16000/1a=imageattr:114 octet-align=0a=fmtp:114 octet-align=0a=rtpmap:115 AMR-WB/16000/1a=imageattr:115 octet-align=1a=fmtp:115 octet-align=1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:101 0-16a=pcfg:1 t=1a=sendrecva=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audio INVITE from the server (after rtpengine)------------------------------------------------------ SIP/2.0 200 OKVia: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rport=37520;received=2.221.73.230From: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>;tag=0ea7681ffd64b0ffcde1bff3393fd505.ae3aCall-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContact: <sip:X.X.X.X:443;transport=tcp>Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 1965 --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=imageattr:114 octet-align=0a=imageattr:115 octet-align=1a=pcfg:1 t=1a=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audioa=rtpmap:114 AMR-WB/16000/1a=rtpmap:115 AMR-WB/16000/1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:114 octet-align=0a=fmtp:115 octet-align=1a=fmtp:101 0-16a=sendrecva=ptime:20a=ice-ufrag:kzgbFAnua=ice-pwd:hnEu9EwTrdIZnTWhYbxtmdugvra=candidate:dzkwmPcB28NPHfRA 1 UDP 16777215 X.X.X.X 30366 typ relay raddr X.X.X.X rport 30366a=candidate:dzkwmPcB28NPHfRA 2 UDP 16777214 X.X.X.X 30367 typ relay raddr X.X.X.X rport 30367a=sendrecva=ice-ufrag:uDE9wbvYa=ice-pwd:dx4CcIjqUNFfnRPJEF8GjatXdU

3 3

Config to control TCP connection reuse
by R, Rajkumar (Raj) 22 Apr '19

22 Apr '19

Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

1 0

Best tutorial for beginners kamailio
by Gaurav Bmotra 22 Apr '19

22 Apr '19

Hi Plz sujjest best kamailio tutorial or any source where a beginner can start kamailio.. Thanks

2 1

push to talk over cellular application server and related media server implementation
by Thierry Luo 21 Apr '19

21 Apr '19

Hi All, I don't know if this is the appropriate place to ask the following questions: 1. Anyone please recommend open-source implementation of PoC (Push-to-talk Over Cellular) application server? Kamailio can do this? 2. Anyone please recommend open-source implementation of PoC related media server/replicator? Thierry Luo

3 2

Re: [SR-Users] authentication using auth_db
by arish haque 21 Apr '19

21 Apr '19

Hi David, This can solve the issue but depth(records) of that view would be too much. The architecture is like this: One kamailio fronting multiple asterisk. Each of asterisk boxes have their Specific users databases. Thus, I'm trying to create db connections inside request route or completely write my own authentication service if I get the user's Password. So, that would be great if I would be able to pass the db_url to auth_db module in request route. Let me know what you think. Thanks, Arish

2 1

Re: [SR-Users] authentication using auth_db
by arish haque 20 Apr '19

20 Apr '19

Hi David, Thanks for the reply. I checked out sqlops module, this will solve the problem only when I get the authentication password (secret) of the user inside request route. For this I tried checking kamailio pseudovars but haven't found one. Therefore, I am sticking with auth_db module.If this is possible by any other way do let me know. thanks, Arish

2 1

Kamailio, TLS, and LetsEncrypt
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i am trying to configure LetsEncrypt (tls) for kamailio sing following link https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ but unable to start kamailio ,i m getting these error ------->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Apr 19 11:23:57 ubuntufresh systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) Apr 19 11:23:58 ubuntufresh kamailio[5365]: Listening on Apr 19 11:23:58 ubuntufresh kamailio[5365]: Aliases: Apr 19 11:23:58 ubuntufresh kernel: kamailio[5378]: segfault at 0 ip 00007f8d6fdac646 sp 00007ffe8e2337f8 error 4 in libc-2.27.so [7f8d6fcfb000+1e7000] Apr 19 11:23:58 ubuntufresh kamailio[5365]: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Control process exited, code=exited status=255 Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Failed with result 'exit-code'. Apr 19 11:23:58 ubuntufresh systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<------------------------------------ plz help -- *Regards:* Gaurav Kumar

3 5

auth by IP
by Yu Boot 19 Apr '19

19 Apr '19

Following code snippet from default kamailio.cfg never gives 403 if you smart enough to set "fromdomain" parameter on Asterisk to Kamailio's IP. How to fix it? I want password-based registration (which is OK now) and permit calls via Kamailio only from permitted IPs. # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; }

2 4

TLS implementation for kamailio
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to implement TLs for kamailio plz help -- *Regards:* Gaurav Kumar

3 2

Microsoft Direct Routing Assistance And Kamailio Integration
by Phillip Cooper 18 Apr '19

18 Apr '19

Hi All, I am looking for some assistance on setting up Microsoft Direct Routing with Kamailio as an SBC. I have a partial config working, I have rewrote the headers going to Direct routing and can place calls in to and out of Direct routing. I have problems with Call hang ups and choppy audio only on the audio stream from Direct Routing. 1) place a call in to direct routing, Hang the call up from the caller and Direct routing hangs up the call. But If you place a call in to Direct routing and hang up the call from direct routing the direct routing hangs up but the call is still active on the caller. IF you call from direct routing through kamailio to a callee and hang up from direct routing the call is hung up correctly. 2) on some calls made to or from direct routing, the audio stream coming from direct routing is sometimes choppy. Its always the audio from direct routing never the audio going to direct routing. My set up: Asterisk <-> Kamailio <-> Direct Routing SIP: Asterisk <-> Kamailio <-> direct routing RTP: Asterisk <-> Direct Routing Kamailio is rewriting the R-URI, TO, FROM, Contact headers going to Direct routing, as Microsoft need the sbc hostname to be in the contact header in a particular format. I am looking for help on the above or if someone would be kind enough to provide a working config that would be awesome! :) Many Thanks in advance for assistance. :) Phil --- This email has been checked for viruses by AVG. https://www.avg.com

1 0

authentication using auth_db module
by arish haque 18 Apr '19

18 Apr '19

Hi all, I'm trying to establish mysql database connection inside request route (at runtime) based on the type of user, as credentials are stored in multiple databases (may be different servers) for the users. Therefore, I am trying to find out that is there anyway to connect to mysql dynamically in request route or pass 'db_url' parameter dynamically to auth_db module. Hope this explains the requirement. Looking for suggestions. With Regards, Arish

2 1

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sr-users