sr-users July 2019

sr-users@lists.kamailio.org

68 participants
74 discussions

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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sr-users July 2019