- sr-users - kamailio.org

Parallel forking and rtpengine handling
by Sebastian Damm 24 Jan '20

24 Jan '20

Hi, our scenario is the following: We have two clients registered to our Kamailio server, one with a TLS capable phone, one via websocket. Now, when a call comes in, the call is forked and is sent out to both clients. rtpengine handling is done in the branch route, so there are two offers, and we use the "via-branch" parameter. Now, when one branch answers the call, what happens to the other branch? I there a way to delete the other branch? How and in which route? Or does Kamailio do this automatically? I'm happy for a hint. Regards, Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH

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Re: [SR-Users] TLS verification error LetsEncrypt
by Social Boh 24 Jan '20

24 Jan '20

I'm not sure but with let's encrypt you can create only server certificate, not client certificate so you can't require and verify client certificate. Regards --- I'm SoCIaL, MayBe El 24/01/2020 a las 09:01, Bugaian A. Vitalie escribió: > Ok, thanks. > > But my question is still about why verification fails/or what should > be chked to make it work. Not how to disable it. > > Thanks. > > Vitalie. > > On Fri, Jan 24, 2020 at 2:54 PM Social Boh <social(a)bohboh.info > <mailto:social@bohboh.info>> wrote: > > Hello, > > changing: > > [client:default] > #method = TLSv1.2+ > verify_certificate = yes > > require_certificate = yes > > with > > [client:default] > #method = TLSv1.2+ > verify_certificate = no > require_certificate = no > > --- > I'm SoCIaL, MayBe > > El 24/01/2020 a las 08:46, Bugaian A. Vitalie escribió: >> Hello list, >> >> I have tried to setup my tls config tish LetsEncrypt following >> this post: >> >> https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ >> >> My tls config looks like this: >> >> >> [server:default] >> method = TLSv1.2+ >> verify_certificate = no >> require_certificate = no >> private_key = /etc/letsencrypt/live/sbc.example.net-0001/privkey.pem >> certificate = >> /etc/letsencrypt/live/sbc.example.net-0001/fullchain.pem >> ca_list = /etc/letsencrypt/live/sbc.example.net-0001/ca_list.pem >> #ca_list = /usr/local/etc/kamailio/tls/cacert.pem >> #crl = /usr/local/etc/kamailio/tls/crl.pem >> server_name = sbc.example.net <http://sbc.example.net> >> server_id = sbc.example.net <http://sbc.example.net> >> >> #ca_list = /usr/local/etc/fullchain.pem >> #ca_list = /usr/local/etc/kamailio/tls/cacert.pem >> #crl = /usr/local/etc/kamailio/tls/crl.pem >> >> >> # --- >> # This is the default client domain profile. >> # 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:default] >> #method = TLSv1.2+ >> verify_certificate = yes >> require_certificate = yes >> >> =================================== >> My ca_list has all certificates from >> cat /etc/ssl/certs/ca-certificates.crt >> >> /etc/letsencrypt/live/sbcc.example.net/ca_list.pem >> <http://sbcc.example.net/ca_list.pem> >> >> I keep getting certificate validation failed see bellow: >> >> an 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6371]: ERROR: tls [tls_util.h:42]: >> tls_err_ret(): TLS write:error:1416F086:SSL >> routines:tls_process_server_certificate:certificate verify failed >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6371]: ERROR: <core> >> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: >> error reading - c: 0x7f0474421f68 r: 0x7f0474422028 (-1) >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6370]: ERROR: tls [tls_util.h:42]: >> tls_err_ret(): TLS write:error:1416F086:SSL >> routines:tls_process_server_certificate:certificate verify failed >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6370]: ERROR: <core> >> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: >> error reading - c: 0x7f0474401cb8 r: 0x7f0474401d78 (-1) >> >> ===================== >> >> What params should I change or where to look for a solution on >> this one? >> >> Thanks. >> >> Vitalie A. Bugaian. >> >> _______________________________________________ >> 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

TLS verification error LetsEncrypt
by Bugaian A. Vitalie 24 Jan '20

24 Jan '20

Hello list, I have tried to setup my tls config tish LetsEncrypt following this post: https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ My tls config looks like this: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/sbc.example.net-0001/privkey.pem certificate = /etc/letsencrypt/live/sbc.example.net-0001/fullchain.pem ca_list = /etc/letsencrypt/live/sbc.example.net-0001/ca_list.pem #ca_list = /usr/local/etc/kamailio/tls/cacert.pem #crl = /usr/local/etc/kamailio/tls/crl.pem server_name = sbc.example.net server_id = sbc.example.net #ca_list = /usr/local/etc/fullchain.pem #ca_list = /usr/local/etc/kamailio/tls/cacert.pem #crl = /usr/local/etc/kamailio/tls/crl.pem # --- # This is the default client domain profile. # 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:default] #method = TLSv1.2+ verify_certificate = yes require_certificate = yes =================================== My ca_list has all certificates from cat /etc/ssl/certs/ca-certificates.crt >> /etc/letsencrypt/live/ sbcc.example.net/ca_list.pem I keep getting certificate validation failed see bellow: an 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6371]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6371]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f0474421f68 r: 0x7f0474422028 (-1) Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6370]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6370]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f0474401cb8 r: 0x7f0474401d78 (-1) ===================== What params should I change or where to look for a solution on this one? Thanks. Vitalie A. Bugaian.

1 0

Releasing Kamailio v5.2.6
by Henning Westerholt 24 Jan '20

24 Jan '20

Hello, after the release of Kamailio 5.3.2 it is time to also release another minor version for the branch 5.2. I plan to release Kamailio 5.2.6 next Friday, 24th January 2020, if there are no new reports of relevant issues that make us to reconsider and postpone. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Ideas to reject a call due to RTPEngine failure when handling the 200Ok
by Patrick Wakano 23 Jan '20

23 Jan '20

Hello list, Hope you all doing well! I have a setup of Kamailio + RTPEngine (so no B2BUA), and I've come to a situation which I can't really figure out the best decision..... The case is a late SDP negotiation that the 200Ok with the SDP offer fails to engage the RTPEngine (for example no RTPEngine available). Also, the same problem can happen with a normal SDP negotiation, in case the INVITE goes through normally but then when engaging the SDP answer for the 200Ok, the RTPengine answer request fails for any reason... In these cases, I can't generate a proper SDP for the 200OOk so the call will connect normally, but the media path will be broken... So for such cases, I am thinking it is reasonable to fail the call. However, failing a call during the 200Ok stage is tricky, at this point we can't convert the 200Ok to a 50X. I tried and Kamailio doesn't allow and also RFC wise this is a no no for a SIP proxy.... I could completely remove the SDP body and forward the 200Ok. This will force a failed SDP negotiation and leave it for the endpoints the task to terminate the call. Is that acceptable? Has anyone faced such situation? Any idea is much appreciated! Kind regards, Patrick Wakano

2 2

uac_replace_from and uac_auth fails to authenticate.
by Kjeld Flarup 23 Jan '20

23 Jan '20

I have a setup where I have a fallback to a GSM number I look up the GSM number and provider information in a database and sets the headers. dlg_manage(); $du = "sip:" + $dbr(ra=>[0,0]); $tu = "sip:"+$rU+"@"+$dbr(ra=>[0,0]); $ru = "sip:"+$rU+"@"+$dbr(ra=>[0,0]); uac_replace_from("sip:"+$dbr(ra=>[0,1])+"@EXTERNALIP"); After this the call goes to a failure_route to do uac_auth() Now my problem is that this works with the providers Asterisk server. But if the call is send to the providers Kamailio server, authentication is rejected. Removing uac_replace_from makes the call accepted on the Kamailio server The only possible problem I can see is that the first INVITE without authentication, has correct From header. But the second with the nonce and auth, uses the wrong From header. Thus two different From headers in the same SIP dialog. Unfortunately uac_replace_from is not allowed in failure_route, so I could test if this is the problem. Is the two different From headers a problem, and how could that be fixed. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

5 6

ims_ipsec_pcscf mandatory for ims_registrar_pcscf ?
by Hamid Hashmi 23 Jan '20

23 Jan '20

According to Module documentation ims_ipsec_pcscf is not mandatory [cid:9c92ff73-4978-4837-b541-a38e15525140] but mod: ims_registrar_pcscf is giving error on ims_ipsec_pcscf. Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:253]: mod_init(): Successfully bound to TM module Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <ul_bind_ims_usrloc_pcscf> in module ims_usrloc_pcscf [/opt/lib64/kamailio/modules/ims_usrloc_pcscf.so] Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: mod_init(): Successfully bound to PCSCF Usrloc module Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/lib64/kamailio/modules/ims_registrar_pcscf.so) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: cdp_avp [cdp_avp_mod.c:225]: cdp_avp_destroy(): Destroying module cdp_avp Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tcpops [tcpops_mod.c:156]: mod_destroy(): TCP keepalive module unloaded. Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:94]: tm_shutdown(): done Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/shm.c:288]: shm_destroy_manager(): destroying memory manager: q_malloc Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/q_malloc.c:1185]: qm_shm_lock_destroy(): destroying the shared memory lock Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/pkg.c:97]: pkg_destroy_manager(): destroying memory manager: q_malloc There is no option to use ims_registrar_pcscf without ims_ipsec_pcscf. [cid:ec21fdae-7d7e-437c-a7ea-6345f36a1428] Regards Hamid R. Hashmi

2 3

rtpengine relay ip and port in avp
by Pafel 23 Jan '20

23 Jan '20

Hello, I am using kamailio 5.3, rtproxy module and sipwise rtpengine. Is there any way I can get the rewritten / relay address and port by the rtpproxy/rtpengine. In avp or some other variable? Regards, Pavel Siderov

2 2

Param.valueat not returning special chars
by Duarte Rocha 21 Jan '20

21 Jan '20

Hello, I've run into a problem while using parameters parsing tools from Kamailio. When i request the value of a parameter it discards some characters like "" for example. Here is an example : $var(Parameters) = "kamailio=\"version5\";sip=hey"; xerr("String Parameter = $var(Parameters)"); xerr("Parameter on position 1 = $(var(Parameters){param.name,0}) and has value = $(var(Parameters){param.valueat,0})"); This code returns : ERROR: <script>: String Parameter = kamailio="version5";sip=hey ERROR: <script>: Parameter on position 0 = kamailio and has value = version5 Is there anything i can do to get the original string? I'm running Kamailio 5.2.0. Best Regards, Duarte Rocha

1 0

Printing xavp
by Ovidiu Sas 21 Jan '20

21 Jan '20

What's the best way to print the entire content of an xavp? The xavp_params_implode() doesn't print the entire stack of an xavp, only the top layer: The following code produces "c=d;e=f;" $xavp(x=>e) = "f"; $xavp(x[0]=>c) = "d"; $xavp(x[1]=>e) = "z"; $xavp(x[1]=>c) = "y"; xavp_params_implode("x", "$var(out)"); Is there a way to print everything inside $xavp(x)? Thanks, Ovidiu

3 10

Re: [SR-Users] Get destination IP from forwarded invite
by CUMHUR KARAHAN 21 Jan '20

21 Jan '20

Hi Joel, Thank you for your response. In fact, I want to get Request-line URI's parameters of the outgoing INVITE that is leaving the first Kamailio box, briefly. Unfortunately "du - destination URL" or "sndto(name)" pseudo-variables indicate the next hope's information. And in this case, the next hope's information is not same as the ClientB's information. As I tried to tell before, our setup is like that: ClientA -> Kamailio1 -> Kamailio2 -> ClientB I need the request line URI parameters of the invite message sent by Kamailio1 to Kamailio2. And these parameters contain ClientB's ip and port information such as 10.252.180.51:36945 as you can see below: INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE Call-ID: 1wF-vjzGIX ... And, your question: no, we don't use the dispatcher mechanism. We just check the DB to find out Kamailio server which the number registered into and send the messages to this server basically. Thanks for all... -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of sr-users-request(a)lists.kamailio.org Sent: 17 Ocak 2020 Cuma 14:00 To: sr-users(a)lists.kamailio.org Subject: sr-users Digest, Vol 176, Issue 17 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. Get destination IP from forwarded invite (CUMHUR KARAHAN) 2. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 3. Re: Get destination IP from forwarded invite (Joel Serrano) 4. Re: Kamalio + RTP Engine + SIP Client behind NAT (Nuno Miguel Reis) 5. Re: Kamailio/RTC dinner at Fosdem 2020 (Federico Santulli) 6. Re: http_client interface (Daniel-Constantin Mierla) 7. Re: Kamailio stop to process incoming SIP traffic via TCP. (adeykunov) 8. Question about xhttp module kamailio 5.2.x (Jimmy Rodley) 9. Trouble with 302 Redirect (Jim Leahy) 10. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 11. Re: http_client interface (Fred Posner) 12. Re: Trouble with 302 Redirect (Alex Balashov) 13. UCaaS Platform (Dario Vargas) 14. Re: [commercial announcement] (Henning Westerholt) ---------------------------------------------------------------------- Message: 1 Date: Thu, 16 Jan 2020 11:45:15 +0000 From: CUMHUR KARAHAN <cumhur.karahan(a)consultant.turkcell.com.tr> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Get destination IP from forwarded invite Message-ID: <319901a90d17413292ad249eb56df2ec(a)consultant.turkcell.com.tr> Content-Type: text/plain; charset="utf-8" Hi all, I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message. Basically, our system/call flow is like: ClientA -> Kamailio1 -> Kamailio2 -> ClientB Invite from the clientA: tls:212.252.34.165:32980 tls:172.21.193.100:5061 INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0 Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 20 INVITE ... Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message. Invite to the Kamailio2 : udp:172.21.193.100:5060 udp:172.21.193.98:5060 INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE ... When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts. Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue. Thanks for all... S. Cumhur Karahan [http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2…> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/37a3c32b/…> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 92 bytes Desc: image001.gif URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/37a3c32b/…> ------------------------------ Message: 2 Date: Thu, 16 Jan 2020 08:23:21 -0500 From: Richard Fuchs <rfuchs(a)sipwise.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <4686712f-d865-c6c9-11ac-649966b867b1(a)sipwise.com> Content-Type: text/plain; charset="utf-8"; Format="flowed" On 15/01/2020 13.39, Nuno Miguel Reis wrote: > Hi guys. > > I'm replacing a environment where I was using kamailio + freeswitch by > another where I'm adding rtpengine to the mix. > One of the issues I'm having now is when I have a SIP Client behind NAT: > > When I send the INVITE from the SIP Client, the SDP is using the > private LAN IP address + the advertised media port, let's admit it's > 10.10.10.10:5000 <http://10.10.10.10:5000> when the call establishes > with a public server running kamailio + rtpengine, the actual RTP > arrives from the home router public IP on a natted port, let's admit > it's 100.100.100.100:65100 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f100.100.100.100%3a65100&umid=39581E1E-9C53-E505-96A6-E057F22 > BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-193c6dfa2da7d1ac0c > 93d1a002e73781a0a22d9e>, event though, RTPENGINE assumes that RTP > stream is coming from > 100.100.100.100 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f100.100.100.100&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&au > th=cbd2d006abdb2241308511f72807fe0bc6594b4a-9e365df6d282d7bf1e79e91e70 > 73c5a94aed6d82>:5000.I'm using the '--sip-source' with RTPENGINE to > make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information. Rtpengine does this automatically, unless the `asymmetric` flag is used. You should see a log message about a `confirmed peer address` as soon as the first RTP is received on a port. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/483ec5d0/…> ------------------------------ Message: 3 Date: Thu, 16 Jan 2020 07:23:28 -0800 From: Joel Serrano <joel(a)textplus.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Get destination IP from forwarded invite Message-ID: <CAMtXxQmmRb-ieA9T+0ofxAxiOd23w_0nHos=nZOy9-JNRahnfA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hello, If I understand correctly your email, you want the IP/Host of the destination of the outgoing INVITE that is leaving a Kamailio box? First of all, how are you forwarding the request? Are you using dispatcher module? Here are a couple of options: 1- https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#du_-_destinat… (although this depends on your config, it might not be set) 2- https://www.kamailio.org/wiki/cookbooks/5.3.x/core#onsend_route + https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#sndto_name If you can describe a little more how you are doing things we might be able to give you more information. Joel. On Thu, Jan 16, 2020 at 3:53 AM CUMHUR KARAHAN < cumhur.karahan(a)consultant.turkcell.com.tr> wrote: > Hi all, > > > > I’m a newbie for Kamailio and working on a feature needs end-users’ > destination IP in the first transferred/forwarded invite message. > > > > Basically, our system/call flow is like: > > ClientA -> Kamailio1 -> Kamailio2 -> ClientB > > > > Invite from the clientA: > > > > *tls:212.252.34.165:32980 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f212.252.34.165%3a32980&umid=39581E1E-9C53-E505-96A6-E057F22B > D18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-d4b503e38b32fd4dee8 > 59434f30f8c5f3e832496>* > > *tls:172.21.193.100:5061 <http://172.21.193.100:5061>* > > *INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 > <http://sip:905364167881@testsip.tvoip.turkcell.com.tr:443> SIP/2.0* > > *Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport* > > *From: <sip:905368601286@testsip.tvoip.turkcell.com.tr > <sip%3A905368601286(a)testsip.tvoip.turkcell.com.tr>>;tag=WbINy0ypx* > > *To: sip:905364167881@testsip.tvoip.turkcell.com.tr > <sip%3A905364167881(a)testsip.tvoip.turkcell.com.tr>* > > *CSeq: 20 INVITE* > > *…* > > > > Needed Ip (Client B’s IP 10.252.180.51 for this case) is in the INVITE > row of the forwarded message. > > > > Invite to the Kamailio2 : > > > > *udp:172.21.193.100:5060 <http://172.21.193.100:5060>* > > *udp:172.21.193.98:5060 <http://172.21.193.98:5060>* > > *INVITE sip:905364167881@**10.252.180.51**:36945;transport=tls > SIP/2.0* > > *Record-Route: > <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes>* > > *Record-Route: > <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f > 5.f572;nat=yes>* > > *Via: SIP/2.0/UDP > gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0 > 593548557.1;i=f* > > *Via: SIP/2.0/TLS > 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;r > port=32980* > > *From: <sip:905368601286@testsip.tvoip.turkcell.com.tr > <sip%3A905368601286(a)testsip.tvoip.turkcell.com.tr>>;tag=WbINy0ypx* > > *To: sip:905364167881@testsip.tvoip.turkcell.com.tr > <sip%3A905364167881(a)testsip.tvoip.turkcell.com.tr>* > > *CSeq: 21 INVITE* > > *…* > > > > When the first invite message receives, the dialog variables or > sel(msg…) parsed values belong to this first received message, as far > as I understand during the tests. However I need the forwarded invite message’s variables. > Even after t_relay in RELAY route, it looks the message I have is the > first one. So, I cannot reach this later IP in the forwarded invite > message. As far as I understand, bear with me here please, this IP > manipulation executes in t_relay function and in the C side of > Kamailio. So, I cannot find a way to do that in the our native scripts. > > > > Is there any way to get it without diving C code’s abyss? Do you have > any idea about it? I would appreciate it, if I can get any help or any clue. > > > > Thanks for all... > > > > S. Cumhur Karahan > > > > > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2fturkcell.li%2fncfYh&umid=39581E1E-9C53-E505-96A6-E057F22BD18 > D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-bbcefdc4443d64fef54812 > 6bd3c5f932d722a0a4> > > Bu elektronik posta ve onunla iletilen butun dosyalar sadece > gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel > kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu > elektronik postanin icerigini aciklamaniz, kopyalamaniz, > yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. > > TURKCELL bu mesajin icerdigi bilgilerin doğruluğu veya eksiksiz oldugu > konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin > ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve > saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca > gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir > > Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. > ------------------------------ > > This e-mail and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you are not the intended recipient you are hereby > notified that any dissemination, forwarding, copying or use of any of > the information is strictly prohibited, and the e-mail should > immediately be deleted. > > TURKCELL makes no warranty as to the accuracy or completeness of any > information contained in this message and hereby excludes any > liability of any kind for the information contained therein or for the > information transmission, reception, storage or use of such in any way > whatsoever. The opinions expressed in this message belong to sender > alone and may not necessarily reflect the opinions of TURKCELL. > > This e-mail has been scanned for all known computer viruses. > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/682b8d5d/…> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 92 bytes Desc: not available URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/682b8d5d/…> ------------------------------ Message: 4 Date: Thu, 16 Jan 2020 17:29:53 +0000 From: Nuno Miguel Reis <nmreis(a)student.dei.uc.pt> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <CAGTw3V7Y8pHD+a8psPL7Tah=HTgrmvVRPhPBKB9VCuoujX_Nnw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi again. Thanks for all the help and suggestions. I realized the issue happens if using kernel forwarding only. If I change rtpengine to start at userspace without the kernel module enabled everything works fine as expected. Do you have any hints on why this could be happening with the kernel module? I'm running rtpengine like this: usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 --listen-ng= 127.0.0.1:2223 --tos=184 --sip-source kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 --listen-ng=127.0.0.1:2223 --tos=184 --no-fallback --sip-source when using kernel module, I'm also setting this rules on chain INPUT under iptables: -A INPUT -p udp -m udp --dport 5000:16383 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 16384:32768 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 49152:65535 -j RTPENGINE --id 0 Looking forward to hear from you guys. Regards, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra On Thu, Jan 16, 2020 at 1:23 PM Richard Fuchs <rfuchs(a)sipwise.com> wrote: > On 15/01/2020 13.39, Nuno Miguel Reis wrote: > > Hi guys. > > I'm replacing a environment where I was using kamailio + freeswitch by > another where I'm adding rtpengine to the mix. > One of the issues I'm having now is when I have a SIP Client behind NAT: > > When I send the INVITE from the SIP Client, the SDP is using the > private LAN IP address + the advertised media port, let's admit it's > 10.10.10.10:5000 when the call establishes with a public server > running kamailio + rtpengine, the actual RTP arrives from the home > router public IP on a natted port, let's admit it's > 100.100.100.100:65100, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. > I'm using the '--sip-source' with RTPENGINE to make it use the > received ip address instead of the private IP coming in the SDP but > I'm not finding anything to make RTPENGINE adapt an start sending the > RTP FLOW to the port where he starts receiving the RTP flow from, > discarding the SDP media port information. > > Rtpengine does this automatically, unless the `asymmetric` flag is used. > You should see a log message about a `confirmed peer address` as soon > as the first RTP is received on a port. > > Cheers > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/d61a6989/…> ------------------------------ Message: 5 Date: Thu, 16 Jan 2020 20:45:04 +0100 From: Federico Santulli <kam.list(a)nhm.it> To: miconda(a)gmail.com, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: "business(a)lists.kamailio.org" <business(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamailio/RTC dinner at Fosdem 2020 Message-ID: <54CDD921-48D2-47F0-935E-0B181F037774(a)nhm.it> Content-Type: text/plain; charset=us-ascii Hi Daniel, We would like to join the dinner, 3 people from US. Kind regards. Federico Santulli NHM - S.R.L. Via Raffaello Sanzio, 88 81031 Aversa (CE) Italy > Il giorno 9 gen 2020, alle ore 09:52, Daniel-Constantin Mierla <miconda(a)gmail.com> ha scritto: > > Hello, > > Fosdem 2020 is approaching, there will be a talk from Henning about > Kamailio, many other friends and related projects are presenting in > the RTC Devroom (Giacomo Vacca and Federico Cabiddu, Wazo, Asterisk, > Homer, Janus, Jitsi, CGRateS, Linphone, ...): > > * https://fosdem.org/2020/schedule/track/real_time_communications/ > > I plan to be at the event and wondering if there are enough interested > participants for having the traditional Kamailio/RTC dinner. If yes, > Torrey will help to book a place to accommodate us (first choice could > be the same restaurant like the last year, if available). > > Reply if you want to join the dinner and say how many other people are > joining you. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- > www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, > 2020, in Berlin -- > https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www. > kamailioworld.com&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d > 006abdb2241308511f72807fe0bc6594b4a-ac4b2b5016612d36d9bf3bf438b6b436b1 > f1ae8e > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ Message: 6 Date: Thu, 16 Jan 2020 22:56:03 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Fred Posner <fred(a)palner.com> Subject: Re: [SR-Users] http_client interface Message-ID: <4ab3422e-c3d6-7d85-fb3e-53b09bfa6c1a(a)gmail.com> Content-Type: text/plain; charset=utf-8 The libcurl seems to offer an option for that as well: - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html So it can be a small patch to the module and support it in the future. If you look for a solution on current stable, maybe one of the embedded scripting languages can help, you can execute inline a small script in Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the http client from lua to do more complex http api queries (e.g., with specific headers, ...). Cheers, Daniel On 13.01.20 17:00, Fred Posner wrote: > I'd never had a reason to before, but on a deployment I need http > messages to go out a specific interface. Curl has an --interface > option, but I don't see how to do the same with http_client. Unless > I'm missing something... > > Anyone ever do something like this? > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www.kamai… ------------------------------ Message: 7 Date: Thu, 16 Jan 2020 09:49:40 -0700 (MST) From: adeykunov <adeykunov(a)intermedia.net> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Kamailio stop to process incoming SIP traffic via TCP. Message-ID: <1579193380553-0.post(a)n5.nabble.com> Content-Type: text/plain; charset=us-ascii Daniel, Got the same issue on 5.3.1 with openssl1.1, debian9. After 3 working days of tests (about ~30-50 wss clients), suddenly we've got a lot of connections stucked in CLOSE_WAIT state. Kamailio called sig_alarm_abort() when we try to reboot. Thanks, Andrey -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html ------------------------------ Message: 8 Date: Thu, 16 Jan 2020 22:02:40 +0000 (UTC) From: Jimmy Rodley <jimmyrodley(a)yahoo.com> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question about xhttp module kamailio 5.2.x Message-ID: <443353431.8467378.1579212160245(a)mail.yahoo.com> Content-Type: text/plain; charset="utf-8" Hi, I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations. The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the request is handled in an asynchronous manner. However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports. https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm104941… I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ? Thanks,Rupesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/f581f9a0/…> ------------------------------ Message: 9 Date: Thu, 16 Jan 2020 15:54:54 -0500 From: Jim Leahy <rickfromsunnyvale(a)gmail.com> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] Trouble with 302 Redirect Message-ID: <CAH3xi0-AEEj_bwLYd4OzSH96Cz8=qyrdsNWTjWnf5Sm1efrvhg(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config: route { # drop some things right away route(DROP); # perform sanity check route(SANITYCHECK); # handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; } # handle invites if(is_method("INVITE")) { route("INVITE"); exit; } route(RELAY); } route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route(); # if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; } onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } } I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily"); But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1 Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily"); But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/66fc8ba2/…> ------------------------------ Message: 10 Date: Thu, 16 Jan 2020 17:10:06 -0500 From: Richard Fuchs <rfuchs(a)sipwise.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <0b704a3c-afc4-3ca0-8d8b-38e2d70bcae9(a)sipwise.com> Content-Type: text/plain; charset="utf-8"; Format="flowed" On 16/01/2020 12.29, Nuno Miguel Reis wrote: > Hi again. > > Thanks for all the help and suggestions. I realized the issue happens > if using kernel forwarding only. If I change rtpengine to start at > userspace without the kernel module enabled everything works fine as > expected. > Do you have any hints on why this could be happening with the kernel > module? > > I'm running rtpengine like this: > > usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 > --listen-ng=127.0.0.1:2223 <http://127.0.0.1:2223> --tos=184 --sip-source > > kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 > --listen-ng=127.0.0.1:2223 <http://127.0.0.1:2223> --tos=184 > --no-fallback --sip-source > The kernel module receives its instructions from the userspace daemon, so there's no reason there should be a difference. Can you post 1) logs and 2) the flags you use for your offers/answers? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/70d3b6de/…> ------------------------------ Message: 11 Date: Thu, 16 Jan 2020 17:11:23 -0500 From: Fred Posner <fred(a)palner.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] http_client interface Message-ID: <9808e784-3d56-d100-4c46-b136cc09ef08(a)palner.com> Content-Type: text/plain; charset=utf-8 On 1/16/20 4:56 PM, Daniel-Constantin Mierla wrote: > The libcurl seems to offer an option for that as well: > > - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html > > So it can be a small patch to the module and support it in the future. > > If you look for a solution on current stable, maybe one of the embedded > scripting languages can help, you can execute inline a small script in > Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the > http client from lua to do more complex http api queries (e.g., with > specific headers, ...). > > Cheers, > Daniel > Thank you Daniel, I implemented a work around (little proxy in go) but would love if the option was added for the future. Fred Posner fred(a)qxork.com https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Direct/SMS: +1 (336) 439-3733 Need Fred? Call Fred. 336-HEY-FRED Matrix: @fred:matrix.lod.com ------------------------------ Message: 12 Date: Thu, 16 Jan 2020 17:31:43 -0500 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Trouble with 302 Redirect Message-ID: <20200116223143.GA857@typhoon> Content-Type: text/plain; charset=us-ascii Try doing this in a failure_route[]; onreply_route[] isn't really meant to transform replies, just to passively observe or drop them. Also, you'll want to use append_to_reply(): https://kamailio.org/docs/modules/5.3.x/modules/textops.html#textops.f.appe… Note that the conditions that give rise to the invocation of a failure_route are broader and somewhat more abstract than any given SIP reply -- e.g. transaction reply timeouts. So, you would want to structure your failure_route like this: failure_route[REDIRECT_HANDLE] { if(t_is_canceled()) exit; if(t_branch_timeout()) { # Handle the timeout case... exit; } # Use $T_rpl to access reply code ($rs) and reply reason if needed # ($rs), per: # # https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#t_rpl_pv if($T_rpl($rs) == 486) { append_to_reply("Contact: <sip:...>\r\n"); t_reply("302", "Moved Temporarily"); exit; } # Some other case. } -- Alex On Thu, Jan 16, 2020 at 03:54:54PM -0500, Jim Leahy wrote: > Hi All, > I'm having some issues trying to properly send a '302 Moved Temporarily' > reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 > status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server > when UAC2 is in DND. > Here are the pertinent parts of my config: > > route { > # drop some things right away > route(DROP); > > # perform sanity check > route(SANITYCHECK); > > # handle registrations > if(is_method("REGISTER")) { > route(REGISTRATION); > exit; > } > > # handle invites > if(is_method("INVITE")) { > route("INVITE"); > exit; > } > > route(RELAY); > } > > route[INVITE] { > # add this proxy to the record-route so it stays in the loop > record_route(); > > # if we can find this user in the location DB... > if(lookup("location")) { > # relay the invite to the new dURI > t_on_failure("FOURDIGITFAIL"); > t_on_reply("FOURDIGITREPLY"); > route(RELAY); > # go back to the main route > return; > } > > onreply_route[FOURDIGITREPLY] { > if(t_check_status("4[0-9]{2}")) { > t_reply("302", "Moved Temporarily"); > } > } > > I'm having issues with the t_reply in the onreply_route. It successfully > sends the reply, but I can't seem to set the Contact header properly. I'm > trying to set it to the location of the user's mailbox on the voicemail > sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: > remove_hf("Contact"); > insert_hf("Contact: sip:1000@vmpbx.domain.com"); > t_reply("302", "Moved Temporarily"); > > But as I understand it, that just changes the header in the reply from > UAC2, not the reply I'm sending to UAC1 > > Then I tried: > append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); > t_reply("302", "Moved Temporarily"); > > But I can't use append_to_reply in an onreply block. > How can I change the Contact header in the reply that's sent? Am I just > going about this the wrong way? Any help would be appreciated. Thanks > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2…, https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2… ------------------------------ Message: 13 Date: Fri, 17 Jan 2020 00:58:49 -0500 From: Dario Vargas <dariovargas.gcs(a)gmail.com> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] UCaaS Platform Message-ID: <CAHqy4U_jecJALyvoZM6p6oGB5VyC_kZN=Sj3_vqjEhkO-tmORw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Take the LEAD in Unified Communications as a Service! We can help you get started in the next Telco Disruptive Business. White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to offer all of the below products under your own brand - Hosted Contact Center (CCaaS) Hosted Unified Communications (UCaaS) Hosted PBX (Full Feature Set) SIP Trunk Ai Chat Bots Caller Profile Solutions (Display your clients' corporate identity on cell phones) to improve phone answer ratio. Video Surveillance (VSaaS integrated with AWS) Web Conferencing Virtual FAX Business SMS Software developed in the USA to have all the routing features required to connect with US and Canadian carriers for inbound and outbound (LRN, LNP, TF, SMS, Push, Video etc). Use our existing contracts and pricing for telco services or BYOC (bring your own carrier model). Alternately we can provide you with introductions to the carriers of choice below to get started - Nuwave.com - Number 1 in TF 8XX Inbound Voipinvite.com - 1000s of MPS and CPS for SMS and Voice Broadcast Voxbone.com - World Wide DiDs - Low Monthly minimum Peerlessnetwork.com - Outbound / Inbound Business calling Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum volumes required Verizon.com - Outbound and Inbound enterprise business calls - minimum volumes required Dario Vargas info(a)dnl-switch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200117/943422d8/…> ------------------------------ Message: 14 Date: Fri, 17 Jan 2020 07:46:13 +0000 From: Henning Westerholt <hw(a)skalatan.de> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Dario Vargas <dariovargas.gcs(a)gmail.com> Subject: Re: [SR-Users] [commercial announcement] Message-ID: <2dcadf18-240a-1eb4-4a46-89d1d84c6007(a)skalatan.de> Content-Type: text/plain; charset="utf-8" Hello Dario, this type of content is not appropriate for the sr-users lists, please post it here. Please use our dedicated business list for this kind of commercial announcements. https://lists.kamailio.org/cgi-bin/mailman/listinfo/business Regards, Henning Am 17.01.20 um 06:58 schrieb Dario Vargas: commercial announcement _______________________________________________ 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://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Kamailio services - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Kamailio Merchandising - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200117/fa884666/…> ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 176, Issue 17 *****************************************

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TLS Profiles : server_name and server_id query
by mahesh b 21 Jan '20

21 Jan '20

Hi All, Am using kamailio 5.1.9 version >From http://www.kamailio.org/docs/modules/5.1.x/modules/tls.html , i understand " The value for server_id can be any string, being used to match TLS client config profile, overriding the match on ip:port and server_name. " Which suggests that the server_name and server_id can be different for a tls profile IN my tls.cfg i have [client:default] verify_certificate = no require_certificate = no [client:10.211.160.172:5071] method = TLSv1.2 verify_certificate = yes require_certificate = yes private_key = /ssl/sip/profile1/server.key certificate = /ssl/sip/profile1/server.crt ca_list = /ssl/sip/profile1/ca.crt cipher_list = RSA verify_depth = 9 server_id = profile1 server_name = btip.176.com And in sar.cfg i am doing, before sending the request out $xavp(tls=>server_name)="btip.176.com"; $xavp(tls[0]=>server_id)="profile1"; But if i see the logs , its picking client default profile for tls handshake, Am i missing anything in tls.cfg or sar.cfg ? if i see the code in tls_lookup_cfg() while (p) { if(srvid && srvid->len>0) { LM_DBG("comparing addr: [%s:%d] [%s:%d] -- id: [%.*s] [%.*s]\n", ip_addr2a(&p->ip), p->port, ip_addr2a(ip), port, p->server_id.len, ZSW(p->server_id.s), srvid->len, ZSW(srvid->s)); if(p->server_id.s && p->server_id.len==srvid->len && strncasecmp(p->server_name.s, srvid->s, srvid->len)==0) { LM_DBG("TLS config found by server id\n"); return p; } .... Why is that there is a strncasecmp between p->server_name.s and srvid->s ?? the strncasecmp should be between p->server_id.s and srvid->s right ? Regards, Mahesh.B

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Re: [SR-Users] [EXT] Re: Kamailio 5.2.2 IMS -enum_query question
by Woscek, Martin W. 21 Jan '20

21 Jan '20

Hi Henning, It turned out that I didn't know how an e.164 lookup is initiated particularly the required format of the calling number. Reading through the source code, the function call: is_e164() showed I was missing the leading + char. I also noticed that I missed the LM_ERR in my log files eg: Jan 10 10:50:11 imskamailio /usr/local/sbin/kamailio[24552]: ERROR: enum [enum.c:840]: enum_query(): R-URI user '<1234567>' is not an E164 number In our case we are looking at standing up a ENUM server that can be queried for specific user characteristics to complete a call, so now we append the '+' when we want to check a caller or callee: enum_pv_query("+$rU", "enumsvc.e164.arpa.", "sip+video") enum_pv_query("+$fU", " enumsvc.e164.arpa", "+sip+video") All is good now. Thanks! Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Saturday, January 18, 2020 6:47 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 IMS -enum_query question Hello Martin, do you see any errors in the log file during the time that you execute the enum_query? Have you tried to test it with activated debug logging? Cheers, Henning Am 10.01.20 um 17:57 schrieb Woscek, Martin W.: Hi, I am trying to invoke unum lookups from calls to enum_query() ,is_from_user_enum() and enum_pv_query(). However we are not seeing any enum lookups over the wire using wireshark. In the Kamailio.cfg the following are set: loadmodule "enum" modparam("enum", "domain_suffix", "mydnsserver.org.") modparam("enum", "i_enum_suffix", " mydnsserver.org.") sample use in our Kamailio.cfg file: if(!enum_query("mydnsserver.org.")) { xlog("....); #do something } else { Xlog("....") #do something else } I do see NAPTR lookups being done by the other Kamailio IMS processes initially when it starts up so I know it has connectivity to the DNS server, but none for the invoked enum queries. We are running Kamailio 5.2.2. Thanks, Martin _______________________________________________ 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://gilawa.com Kamailio Merchandising - https://skalatan.de/merchandising/

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Re: [SR-Users] Kamailio service error
by Voip support 21 Jan '20

21 Jan '20

I think you need to capture w tcpdump on kamailio port and then check what packet is causing this. tcmdump -f "port 5060" -w file.pcap Then open file in wireshark and try figure out ( i think not valid sip packet coming on 5060 port). wt., 21 sty 2020 o 11:26 Mike from ApoloCore <mike(a)apolocore.com> napisał(a): > I think you need to capture w tcpdump on kamailio port and then check what > packet is causing this. > > tcmdump -f "port 5060" -w file.pcap > > Then open file in wireshark and try figure out ( i think not valid sip > packet coming on 5060 port). > > wt., 14 sty 2020 o 08:54 Alex Balashov <abalashov(a)evaristesys.com> > napisał(a): > >> It would appear that an incoming SIP message has one or more >> grammatically invalid headers that confuse the parser? >> >> — >> Sent from mobile, with due apologies for brevity and errors. >> >> > On Jan 14, 2020, at 2:37 AM, Ratanraj Singh <ratanraj.singh2(a)gmail.com> >> wrote: >> > >> > >> > Hi All, >> > I am troubling with kamailio service error. Can someone help on this >> issue? >> > >> > >> > <image.png> >> > >> > _______________________________________________ >> > Kamailio (SER) - Users Mailing List >> > sr-users(a)lists.kamailio.org >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> _______________________________________________ >> 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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Possible bug in Presence module !?
by SamyGo 21 Jan '20

21 Jan '20

Hi, I'm experiencing a strange behavior with the Kamailio presence module. I've a Polycom phone which takes the the XML tag "version" very seriously and so as soon it SUBSCRIBEs to a parking lot a NOTIFY is sent from server with "version=1" - When the Parking lot/extension gets busy it receives the second NOTIFY with "version=1" whereas it should've been 2. So it fails to light up the BLF icons. Now, any further notify correctly increments the version tag and Polycom blf works fine until its SUBSCRIBE expires and the first cycle of NOTIFY is exchanged. I can tell its a bug because for each NOTIFY the *event_route[tm:local-request]* prints the attribute $subs(version) and it correctly shows the sequence 1,2,3...onwards, However the sngrep/network trace tells it differently, the version shows up as 1,1,2,3...onwards *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 22:20:40 Jan 18 2020 with gcc 4.9.2 *MODULES:*#!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_mwi.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_reginfo.so" loadmodule "pua.so" loadmodule "pua_rpc.so" loadmodule "pua_dialoginfo.so" modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", "sip:MY_HOST_IP:MY_HOST_PORT") modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 1) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "db_update_period", 60) modparam("presence", "db_table_lock_type", 0) modparam("presence", "local_log_level", 3) modparam("presence", "notifier_processes", 10) modparam("presence", "force_delete", 0) modparam("presence", "subs_db_mode", 2) modparam("presence", "db_table_lock_type", 0) modparam("presence", "expires_offset", 60) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 30) modparam("presence", "publ_cache", 0) modparam("presence", "min_expires_action", PRESENCE_MIN_EXPIRES_ACTION) modparam("presence", "min_expires", PRESENCE_MIN_EXPIRES) modparam("presence", "max_expires", PRESENCE_MAX_EXPIRES) modparam("presence", "sip_uri_match", 1) modparam("presence", "waitn_time", 1) modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:MY_HOST_IP:MY_HOST_PORT") modparam("presence_xml", "force_dummy_presence", 1) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "db_url", DBURL) modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence_dialoginfo", "force_dummy_dialog", 1) #!endif Looking to find some answers, Thanks, Sammy

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Kamalio + RTP Engine + SIP Client behind NAT
by Nuno Miguel Reis 20 Jan '20

20 Jan '20

Hi guys. I'm replacing a environment where I was using kamailio + freeswitch by another where I'm adding rtpengine to the mix. One of the issues I'm having now is when I have a SIP Client behind NAT: When I send the INVITE from the SIP Client, the SDP is using the private LAN IP address + the advertised media port, let's admit it's 10.10.10.10:5000 when the call establishes with a public server running kamailio + rtpengine, the actual RTP arrives from the home router public IP on a natted port, let's admit it's 100.100.100.100:65100, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. I'm using the '--sip-source' with RTPENGINE to make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information. In my previous setup with freeswitch alone, freeswitch did this transparently. How do you guys solve this issue in RTPENGINE? Regards, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

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rfc: moving lib srutils to core
by Daniel-Constantin Mierla 20 Jan '20

20 Jan '20

Hello, during the last developers meeting in Dusseldorf we had some discussions about simplifying the building process and internal dependencies/libraries. The src/lib/srutils contains now mainly the uuid and json functions, which are used by very common modules such as registrar/usrloc, tm, dialog, jsonrpcs, dmq*, ... There is a part related to the time recurrence matching, used by drouting, tmrec, but the code is not that significant in size, so I am planning to relocate the lib code to the core and get rid of dependencies to internal libsrutils to many modules. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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Ann: Registration to Kamailio World 2020 is open!
by Daniel-Constantin Mierla 20 Jan '20

20 Jan '20

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (April 27-29, 2020, in Berlin, Germany) is now open! More details are available at: * https://www.kamailioworld.com/k08/registration/ Call for presentations is still open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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Kamailio 5.2.2 IMS -enum_query question
by Woscek, Martin W. 18 Jan '20

18 Jan '20

Hi, I am trying to invoke unum lookups from calls to enum_query() ,is_from_user_enum() and enum_pv_query(). However we are not seeing any enum lookups over the wire using wireshark. In the Kamailio.cfg the following are set: loadmodule "enum" modparam("enum", "domain_suffix", "mydnsserver.org.") modparam("enum", "i_enum_suffix", " mydnsserver.org.") sample use in our Kamailio.cfg file: if(!enum_query("mydnsserver.org.")) { xlog("....); #do something } else { Xlog("....") #do something else } I do see NAPTR lookups being done by the other Kamailio IMS processes initially when it starts up so I know it has connectivity to the DNS server, but none for the invoked enum queries. We are running Kamailio 5.2.2. Thanks, Martin

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Question about xhttp module kamailio 5.2.x
by Jimmy Rodley 18 Jan '20

18 Jan '20

Hi, I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations. The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the request is handled in an asynchronous manner. However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports. https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm104941… I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ? Thanks,Rupesh

2 1

http_client interface
by Fred Posner 17 Jan '20

17 Jan '20

I'd never had a reason to before, but on a deployment I need http messages to go out a specific interface. Curl has an --interface option, but I don't see how to do the same with http_client. Unless I'm missing something... Anyone ever do something like this? -- Fred Posner fred(a)qxork.com https://qxork.com Direct/SMS: +1 (336) 439-3733 Need Fred? Call Fred. 336-HEY-FRED Matrix: @fred:matrix.lod.com

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UCaaS Platform
by Dario Vargas 17 Jan '20

17 Jan '20

Take the LEAD in Unified Communications as a Service! We can help you get started in the next Telco Disruptive Business. White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to offer all of the below products under your own brand - Hosted Contact Center (CCaaS) Hosted Unified Communications (UCaaS) Hosted PBX (Full Feature Set) SIP Trunk Ai Chat Bots Caller Profile Solutions (Display your clients' corporate identity on cell phones) to improve phone answer ratio. Video Surveillance (VSaaS integrated with AWS) Web Conferencing Virtual FAX Business SMS Software developed in the USA to have all the routing features required to connect with US and Canadian carriers for inbound and outbound (LRN, LNP, TF, SMS, Push, Video etc). Use our existing contracts and pricing for telco services or BYOC (bring your own carrier model). Alternately we can provide you with introductions to the carriers of choice below to get started - Nuwave.com - Number 1 in TF 8XX Inbound Voipinvite.com - 1000s of MPS and CPS for SMS and Voice Broadcast Voxbone.com - World Wide DiDs - Low Monthly minimum Peerlessnetwork.com - Outbound / Inbound Business calling Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum volumes required Verizon.com - Outbound and Inbound enterprise business calls - minimum volumes required Dario Vargas info(a)dnl-switch.com

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Kamailio stop to process incoming SIP traffic via TCP.
by Kristijan Vrban 16 Jan '20

16 Jan '20

Hi kamailios, i have a creepy situation with v5.2.1 stable Kamilio. After a day or so, Kamailio stop to process incoming SIP traffic via TCP. The incoming TCP network packages get TCP-ACK from the OS (Debian 9, 4.18.0-15-generic-Linux) but Kamailio does not show any processing for the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via UDP is working just totally fine. When i look via command "netstat -ntp" is see, that the Recv-Q get bigger and bigger. e.g.: Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED 31347/kamailio After Kamailio restart, all is working fine again for a day. We have maybe 10-20 devices online via TCP and low call volume (1-2 call per minute). The only settings for tcp we have is "tcp_delayed_ack=no" How to could we debug this situation? Again, no error, no warings in the log. Just nothing. Kristijan

11 46

Get destination IP from forwarded invite
by CUMHUR KARAHAN 16 Jan '20

16 Jan '20

Hi all, I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message. Basically, our system/call flow is like: ClientA -> Kamailio1 -> Kamailio2 -> ClientB Invite from the clientA: tls:212.252.34.165:32980 tls:172.21.193.100:5061 INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0 Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 20 INVITE ... Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message. Invite to the Kamailio2 : udp:172.21.193.100:5060 udp:172.21.193.98:5060 INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE ... When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts. Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue. Thanks for all... S. Cumhur Karahan [http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] <http://turkcell.li/ncfYh> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses.

2 1

Negative ACK issue
by Michael Broughton 15 Jan '20

15 Jan '20

Hello, Long time Kam/Ser user, first time poster. I'm running into a problem with one of our telco providers when we make a call that ends up being not in service or some other error. In this case our ACK's are not working and the phone line stays open for a period of time until something times out on their end. They claim the issue is that our negative ACK message is dropping one of the Via headers. This is the only case I can find in our setup where Kamailio does this. But it does drop the first Via, which is the first hop in our internal network. I don't understand why this is a problem for them, and I'm still trying to get a reasonable explanation out of them. Technically, I don't see why it would be a problem. This behaviour is not an issue with our other telco providers. Strangely enough, it is also not an issue for this provider when we make the calls over their MPLS network (we are switching to the internet). My question is, can this behaviour be changed in Kamailio somehow? Is there a way for it to keep all the Via headers for negative ACK's? Or, do I just need to poke them harder to fix their issues? My setup: Source -> Kamailio -> Firewall (NAT, SIP ALG) -> Telco Provider I hope I have provided enough information. Thanks! Michael

4 9

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