- sr-users - kamailio.org

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

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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;

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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

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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

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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*

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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

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