- sr-users - kamailio.org

Re: [SR-Users] P-CSCF not starting- Kamailio 5.3
by Henning Westerholt 04 Feb '20

04 Feb '20

Hi Lanvin, (please keep the sr-users list in CC) the modules works on my machine: henning@linux:~/repositories/kamailio> killall kamailio ; ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.188.25 [192.168.188.25]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.188.25 [192.168.188.25]:5060 Aliases: henning@linux:~/repositories/kamailio> ps aux |grep kamailio henning 15948 0.1 0.1 166776 14900 ? S 15:33 0:00 ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d henning 15949 0.0 0.0 166776 4224 ? S 15:33 0:00 ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d [...] Here the diff to the default cfg in git master: diff --git a/etc/kamailio.cfg b/etc/kamailio.cfg index c0e618719..dbc911fb0 100644 --- a/etc/kamailio.cfg +++ b/etc/kamailio.cfg @@ -309,6 +309,14 @@ loadmodule "pike.so" loadmodule "debugger.so" #!endif + +loadmodule "ims_dialog" +loadmodule "ims_usrloc_pcscf" +loadmodule "ims_ipsec_pcscf" +loadmodule "ims_registrar_pcscf" +modparam("ims_dialog", "dlg_flag", 4) + + Cheers, Henning Am 04.02.20 um 11:55 schrieb Lanvin ADON: > > > loadmodule "ims_dialog" > loadmodule "ims_usrloc_pcscf" > #loadmodule "ims_registrar_pcscf" > #!ifdef WITH_IPSEC > loadmodule "ims_ipsec_pcscf" > #!endif > loadmodule "ims_registrar_pcscf" > > > > > Feb 4 11:53:27 pcscf kamailio[17616]: 0(17618) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:27 pcscf kamailio[17616]: 0(17618) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:27 pcscf kamailio[17616]: ERROR: error while initializing > modules > Feb 4 11:53:28 pcscf kamailio[17620]: 0(17622) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:28 pcscf kamailio[17620]: 0(17622) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:28 pcscf kamailio[17620]: ERROR: error while initializing > modules > Feb 4 11:53:28 pcscf kamailio[17623]: 0(17625) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:28 pcscf kamailio[17623]: 0(17625) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:28 pcscf kamailio[17623]: ERROR: error while initializing > modules > Feb 4 11:53:29 pcscf kamailio[17626]: 0(17628) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:29 pcscf kamailio[17626]: 0(17628) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:29 pcscf kamailio[17626]: ERROR: error while initializing > modules > Feb 4 11:53:29 pcscf kamailio[17629]: 0(17631) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:29 pcscf kamailio[17629]: 0(17631) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:29 pcscf kamailio[17629]: ERROR: error while initializing > modules > > > > > > > > > On Tue, Feb 4, 2020 at 11:46 AM Henning Westerholt <hw(a)skalatan.de > <mailto:hw@skalatan.de>> wrote: > > Hello Lanvin, > > try to load it before and see if it starts then. > > Cheers, > > Henning > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com <https://gilawa.com/> > > *From:* Lanvin ADON <lanvin.adon(a)gmail.com > <mailto:lanvin.adon@gmail.com>> > *Sent:* Tuesday, February 4, 2020 11:45 AM > *To:* Henning Westerholt <hw(a)skalatan.de <mailto:hw@skalatan.de>> > *Subject:* Re: [SR-Users] P-CSCF not starting- Kamailio 5.3 > > Hello Henning > > ims_ipsec_pcscf is loaded after. > > loadmodule "ims_dialog" > loadmodule "ims_usrloc_pcscf" > loadmodule "ims_registrar_pcscf" > #!ifdef WITH_IPSEC > loadmodule "ims_ipsec_pcscf" > #!endif > > Thanks > > Lanvin ADON > > On Tue, Feb 4, 2020 at 11:06 AM Henning Westerholt <hw(a)skalatan.de > <mailto:hw@skalatan.de>> wrote: > > Hello, > > just a guess, but do you load the ims_ipsec_pcscf module > before the ims_registrar_pcscf module? > > Cheers, > > Henning > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com <https://gilawa.com/> > > *From:* sr-users <sr-users-bounces(a)lists.kamailio.org > <mailto:sr-users-bounces@lists.kamailio.org>> *On Behalf Of > *Lanvin ADON > *Sent:* Tuesday, February 4, 2020 8:54 AM > *To:* sr-users(a)lists.kamailio.org > <mailto:sr-users@lists.kamailio.org> > *Subject:* [SR-Users] P-CSCF not starting- Kamailio 5.3 > > Hi All, > > I'm installing an IMS core with Kamailio 5.3 release. The > P-CSCF is not starting with the following error message. > > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> > [core/sr_module.c:638]: find_mod_export_record(): export of > <bind_ims_ipsec_pcscf> not found (flags 0) > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: > mod_init(): can't bind ims_ipsec_pcscf > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing > module ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while > initializing modules > > For the configuration file, I'm using the example files in the > source. > > > > -- > > Regards > > Lanvin ADON > > > > -- > > Cordialement, > > Lanvin ADON > > > > -- > > */ > /* > > Cordialement, > > Lanvin ADON -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com Kamailio Merchandising - https://skalatan.de/merchandising/

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Re: [SR-Users] P-CSCF not starting- Kamailio 5.3
by Henning Westerholt 04 Feb '20

04 Feb '20

Hello Lanvin, try to load it before and see if it starts then. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Lanvin ADON <lanvin.adon(a)gmail.com> Sent: Tuesday, February 4, 2020 11:45 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hello Henning ims_ipsec_pcscf is loaded after. loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" loadmodule "ims_registrar_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif Thanks Lanvin ADON On Tue, Feb 4, 2020 at 11:06 AM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, just a guess, but do you load the ims_ipsec_pcscf module before the ims_registrar_pcscf module? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Lanvin ADON Sent: Tuesday, February 4, 2020 8:54 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON -- Cordialement, Lanvin ADON

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P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

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parallel forking in combination with rtpengine
by michael.willow＠web.de 04 Feb '20

04 Feb '20

Hi everbody, I encountered an issue with my Kamailio in combination with Rtpengine. I wondered if anybody got a solution or a hint as this issue is only in combination with special parallel forked calls. I sent a call from my SIP gateway via a Kamailio to another SIP provider. The other SIP provider or the B-user seems to use a parallel forking as I receive 183 Session Progress and 180 Ringing with different to-tags. My kamailio is using a Rtpengine to relay the media from my SIP gateway to the other SIP provider. As Kamailio receives two different to-tags that include a SDP body rtpengine_manage makes Rtpengine to change RTP port. Some short example (cut down to the important messages): a-gateway kamailio b-provider | INVITE Port 16890 | INVITE Port 20400 | | ------------------------> | --------------------------> | | | | | 180 to-tag 1 / no SDP | 180 to-tag 1 / no SDP | | <------------------------ | <------------------------- | | | | | 183 to-tag 2 / Port 20408 | 183 to-tag 2 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20400 ----> Port 46298 | | | | | | | | 180 to-tag 2 / Port 20408 | 180 to-tag 0 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | 200 OK to-tag 1 / Port 20408 | 200 OK to-tag 1 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20418 ----> Port 46298 | Kamailio receives INVITE from a-gateway with rtp port 16890 and rtpengine rewrites it with port 20400 to b-provider. b-provider sends 180 ringing from to-tag 1 and no SDP which is just passed through kamailio to a-gateway. b-provider sends 183 progress from to-tag 2 and SDP with port 46298 to kamailio. rtpengines rewrites the sdp port to 20408 which is passed to a-gateway. b-provider start sending RTP via the negotiated ports which is passed correctly to a-gateway. b-provider sends 180 ringing from to-tag 2 with the same SDP. Rewritten and passed to a-gateway. b-provider sends 200 OK on Invite. This 200 Ok is from to-tag 1 but including the same SDP endpoint as to-tag 2 used before. Media from b-provider via rtpengine to a-gateway is passed the same way as before. Media from a-gateway to rtpengine is passed the same way as before. Media from rtpengine is passed with a different source port to b-provider. As this port was not announced in the callflow b-provider blocks the traffic. Any ideas how to fix this issue or force rtpengine to stay on the same port? Thanks Michael

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P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

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Call merging issues on cellular network
by Gaurav Bmotra 03 Feb '20

03 Feb '20

Hi I m using kamailio 5.3 I m trying to merge the call on cellular network but is not working. As on WiFi it is working fine . Any one can help me Thank you

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Kamailio TLS server hello without any client certificate request
by Patrick Murphy 03 Feb '20

03 Feb '20

Hey guys - I'm using the following settings on my Kamailio config: modparam("tls", "low_mem_threshold2", 1024) modparam("tls", "tls_force_run", 1) modparam("tls", "tls_disable_compression", 1) modparam("tls", "ssl_max_send_fragment", 4096) modparam("tls", "renegotiation", 0) modparam("tls", "low_mem_threshold1", 2048) modparam("tls", "require_certificate", 1) When a client attempts to connect to Kamailio, the server hello it receives does *not* contain the client certificate request. And I've noticed that kamailio starts with: tls [tls_domain.c:694]: set_verification(): TLSs<default>: No client certificate required and no checks performed tls [tls_domain.c:320]: fill_missing(): TLSs<default>: require_certificate=0 Complete version output: kamailio 5.1.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. Any hints?

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Dialog module - default_timeout behavior
by Gholamreza Sabery 03 Feb '20

03 Feb '20

Hi, According to the documentation default_timeout is "The default dialog timeout (in seconds), in the absence of a custom value provided in an AVP". There is also a "send_bye" option which can automatically send a BYE message on dialog timeout. Suppose that a user sends an INVITE and the call is successful (200). He talks more than default_timeout seconds (so there is not BYE for more than default_timeout seconds). This means that when the user actually sends BYE, the dialog is already terminated by the dialog module? Regards

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sqlops multiple databases
by Ali Taher 03 Feb '20

03 Feb '20

Hello, Is there a way in sqlops module to use multiple postgres databases based on routing logic ? Here is what I'm using right now: modparam("sqlops","sqlcon","cb=>postgres://postgres:postgres@localhost:5432/db1") Thanks Ali Taher

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Bad request when INVITE relayed to another kamailio
by Zhan Bazarov 31 Jan '20

31 Jan '20

We have Kamailio-cluster via route53(round-robin) some-domain.net we have two kamailio with public IP's phone1 is registered on kam1 phone2 is registered on kam2 when we are calling from phone1 to phone2 callflow looks: phone1 => kam1 => asterisk => kam1 => t_relay(address of second kamailio:5078) => kam2 => phone2 it works perfectly, but in case when we are using polycom as phone2 - we are getting 404 response from polycom... *Invite from second kamailio * 2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.191:5078;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.135:5078;lr> Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135> Contact: <sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1> Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 CSeq: 22619 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Penny" <sip:1015@10.199.240.179> Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463 *Response from POLYCOM * 2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135>;tag=8BC58304-83D9B045 CSeq: 22619 INVITE Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.19:5078;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.235:5078;lr> User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848 Accept-Language: en Content-Length: 0 Any ideas how to fix it? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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$vn vs $var
by Alex Balashov 31 Jan '20

31 Jan '20

Hi, I must have missed the discussion about it somewhere, but what is the inspiration for the new $vn variable container? According to the documentation, the only difference is that it can hold $null values, whereas variables initialised in $var to $null are coerced into an integer value of 0. But why was this a concern? Why is it necessary? Is it not confusing to users? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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rtpengine relay ip and port in avp
by Pafel 31 Jan '20

31 Jan '20

Thanks for the info Yuri! I tried several scenarios but without success. My sip proxy has ipv4 and ipv6 address. The other nodes use ipv4. When there is a request for a ipv6 UAC/customer comming to the proxy the rtpengine/sipwise is rewriting the sdp with the correct ipv6 address of the proxy. So I am trying to get the ipv6 address of the proxy from the sdp (which is always the same ipv6 address) and the port (which is always different). I've enabled modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") And I have in request section: rtpengine_manage($avp(rtpproxy_offer_flags)); set_body("$avp(sdp)", "application/sdp"); xlog("L_INFO", "Request SDP $avp(sdp))\n"); And the problem is that I am always getting the ipv4 address and port but not the rewritten by the proxy ipv6 address and port. Any idea? Regards, Pavel

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Re: [SR-Users] modifying via header from public IP to private IP
by vinod mn 31 Jan '20

31 Jan '20

Thank's david in my kamailio configuration I am using *listen=udp:192.168.131.69:5060 <http://192.168.131.69:5060/> advertise 221.20.20.12:5060 <http://221.20.20.12:5060/>* (as mentioned by you) and when the kamailio sends an invite to sip-gateway I am seeing this sip-gateway ip is *192.168.132.67 * and this gateway don't have any public-ip assigned .. Via: *SIP/2.0/UDP 221.20.20.12:5060 <http://221.20.20.12:5060/>;* branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 I want private ip instead of public ip in the first via -- Thanks and regards Vinod.M.N

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kamailio
by Genildo Nascimento 30 Jan '20

30 Jan '20

Someone managed installed kamailio with IPV6

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modifying via header from public IP to private IP
by vinod mn 30 Jan '20

30 Jan '20

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. Please let me know, is it possible to replace. [image: image.png] the via headers that kamailio is sending to sip-gateway is sip-phone ip:192.168.113.112 Via: SIP/2.0/UDP *221.20.20.12:5060 <http://221.20.20.12:5060>* ;branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 how to modify the first via header to have private-IP instead of public-IP.. I tried all the replace functions but it is not working. -- Thanks and regards Vinod.M.N

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How to make a distinction between sip and tel uri
by Duarte Rocha 29 Jan '20

29 Jan '20

Greetings, I have a client that sends me two P-Asserted-Identity headers, one with SIP URI and one with tel URI. The pseudo variable $ai returns the topmost P-Asserted-Identity header and I only need the one with the SIP URI. How should i proceed ? Is there a way to delete the one with tel URI? the header removal functions i find only work with header name. I could also work with something different than $ai for this scenario. Best Regards,

2 1

Re-write contact header
by egemen ulus 29 Jan '20

29 Jan '20

Hi everyone, I have asterisk farm and kamailio which located front of asterisks farm as a proxy. When someone send a invite from outside to my kamailio (without record route), i forward it to my Asterisk farm. Then My one of Asterisk reply it as 200 ok. Then my kamailio forward this 200 ok to remote side with record route. But in my 200 ok message, contact header contains my Asterisk IP address. That is why remote side tries to send ACK to Asterisk. How to rewrite my contact header as my kamailio (SBC) address ? Actually I tried fixed nated contact and advertise ip but I couldn't make it worked. I am really new on that. Kamailio cfg file is attached. Please help! Egemen Get Outlook for Android<https://aka.ms/ghei36>

3 4

Kamailio TLS
by Himala Gamage 28 Jan '20

28 Jan '20

Hello All, Kamailio newbie here, I'm trying to set up TLS on Kamailio and having some issues, Kamailio Running on Debian 10. This what I get when trying to install TLS apt-get install -y kamailio-tls-modules The following packages have unmet dependencies: kamailio-tls-modules : Depends: libcurl3 (>= 7.16.2) but it is not installable E: Unable to correct problems, you have held broken packages. Can I install the TLS package with libcurl4? If not, any other workaround? Thanks, In advance Himala

2 4

SIP Redirect CANCEL handling
by Ali Taher 28 Jan '20

28 Jan '20

Hello, I'm using Kamailio as sip redirect server where SBC forwards the calls to Kamailio which sends the routing back to SBC in 3xx response. Now, I need to handle CANCEL requests sent from SBC to Kamailio. Here is the scenario: the client sends CANCEL request to SBC which responds with 200 OK and 487 Request Terminated and then sends CANCEL request to Kamailio. I'm not sure how Kamailio should handle the CANCEL request here, should it send only 200 OK back to SBC or also should send 487 request terminated? Regards, Ali Taher

3 7

get first occurrence value of a param in a string
by Ali Taher 28 Jan '20

28 Jan '20

Hello, I need to get the value of first tgrp param in this string : $var(x) = <sip:xxxxxxxxx;tgrp=0001000;trunk-context=xx.xx.xx.xx@xx.xx.xx.xx:5060;transport=UDP;user=phone;tgrp=237> I'm trying $var(y) = $(var(x){param.value,tgrp}); but it's getting the last value of tgrp which is "237>". Noting that first tgrp is not always in the second index , some other parameters can be added to the string. How to get the value of first occurrence of tgrp param ? Regards, Ali Taher

1 0

centos 7 repo is broken
by Aidar Kamalov 27 Jan '20

27 Jan '20

Hello # yum install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.reconn.ru * extras: mirror.docker.ru * nux-dextop: mirror.li.nux.ro * updates: dedic.sh Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.6-7.el7.centos will be installed --> Processing Dependency: /usr/bin/python23 for package: kamailio-5.2.6-7.el7.centos.x86_64 --> Processing Dependency: /usr/bin/python23 for package: kamailio-5.2.6-7.el7.centos.x86_64 --> Finished Dependency Resolution Error: Package: kamailio-5.2.6-7.el7.centos.x86_64 (kamailio) Requires: /usr/bin/python23 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest -- Aydar A. Kamalov

2 1

Kamailio on Docker
by Jack R 27 Jan '20

27 Jan '20

Hi Please advise have you tried deploying Kamailio (just plain SIP server) with mysql on Docker. I was trying the same, using Kamailio & mysql images on https://hub.docker.com however there are two issues that I was facing: 1. I ran Wireshark on server & inside container I am getting 403 Not relaying and SIP client MircoSIP does not connect. Request reaches the server and the container however it gives 403 Not relaying. I thought to print myuri value from Kamailio.cfg, to understand the issue better however then comes the second issue as below: 2. This is a bigger issue: I am not able to generate any logs -no syslogs, no docker logs (very limited logs get generated). Hence not able to troubleshoot anything. Please advise and help. Thanks and Regards, Jack.

2 1

Kamailio/RTC dinner at Fosdem 2020
by Daniel-Constantin Mierla 27 Jan '20

27 Jan '20

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 -- www.kamailioworld.com

6 6

CSeq values in dialog OPTIONS keepalives
by Alex Balashov 24 Jan '20

24 Jan '20

Hi, I'd like to reopen an old thread of discussion: Using Kamailio 4.4.x latest, when I enable dialog OPTIONS keepalives on both sides of a call - ka-dst and ka-src dialog params, that is - I get OPTIONS pings sent to one side with a CSeq value like this: CSeq: 0 OPTIONS and the other side: CSeq: (CSeq of e2e ACK - 1) OPTIONS The 0 CSeq is acceptable to most UAs--at least, among the small number I've tested--and they answer 200 OK, but the second one results in a '500 Server Internal Error'. I have read the prior literature on this: https://lists.kamailio.org/pipermail/sr-users/2012-May/073069.html https://lists.kamailio.org/pipermail/sr-users/2018-April/101096.html and I (mostly) understand the rationale of keeping this mechanism stateless. I tried turning on `track_cseq_updates` to see if it might change the behaviour, but it does not. Anyway, a mechanism that deliberately elicits 500 responses from the endpoint is going to raise some eyebrows and just looks a bit unprofessional. I'm wondering if there is a better way to go. My questions are: 1. While I understand the reasoning here: "it would results de-synchronization of the CSeq values hold in phones themselves (e.g., a BYE created by caller/callee after a keep alive will be with lower cseq than the other side would expect and accept)." Would it be such a problem to use a CSeq value that is (last highest known CSeq observed) + ($RANDOM % LARGE_VALUE)? 2. In the case that `ka-dst` and `ka-src` are both enabled, why is there an inconsistency in the behaviour with respect to the upstream and downstream side (CSeq value of 0 vs. CSeq value of <= CSeq(ACK))? 3. At the risk of inviting some baroque state-keeping that is runtime-dependent, could there be an implementation where the CSeq of genuine in-dialog requests from the UA is modified in-flight by Kamailio, taking advantage of its being in the middle of in-dialog requests, to the appropriate next highest value? I ask this because, if I understood the limited documentation for `track_cseq_updates` correctly, https://kamailio.org/docs/modules/5.3.x/modules/dialog.html#dialog.p.track_… "Enable the callbacks for tracking if CSeq number needs to be updated. It is the case when the INVITE has to be authenticated to downstream provider using uac_auth() from uac module. This is done only for requests in downstream direction. The CSeq difference is stored in $dlg_var(cseq_diff), be sure this variable is not overwritten via config operation." it seems like the door to this method might already be open? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

TLS and TCP port question
by Cyrille Demaret 24 Jan '20

24 Jan '20

Hi, Im trying to implement kamailio as a Teams SBC as described here : <https://skalatan.de/en/blog/kamailio-sbc-teams> https://skalatan.de/en/blog/kamailio-sbc-teams (thanks to the author, I think hes here) Im using Kamailio 5.2.1 and I still have a problem to have the SBC active on the Microsoft side. Im correctly receiving the OPTIONS packet from Microsoft but I have some doubt on my OK reply. Heres the dialog captured in sngrep using the siptrace module : The OPTIONS packet I receive from Microsoft contains a Contact header with a different port as the original packet. So, Kamailio send the 200 OK back to this port. However, when I capture packets using tcpdump, I can see that theres no connection to 52.114.76.76:5061 : 15:01:35.466926 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 1, win 2053, length 0 15:01:35.467383 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 1:189, ack 1, win 2053, length 188 15:01:35.467423 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [.], ack 189, win 237, length 0 15:01:35.478535 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [.], seq 1:2881, ack 189, win 237, length 2880 15:01:35.478771 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 2881:3017, ack 189, win 237, length 136 15:01:35.498028 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 2881, win 2053, length 0 15:01:35.551569 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 3017, win 2052, length 0 15:01:35.765965 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 189:282, ack 3017, win 2052, length 93 15:01:35.767719 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 3017:3259, ack 282, win 237, length 242 15:01:35.800634 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 282:825, ack 3259, win 2051, length 543 15:01:35.802362 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 3259:3656, ack 825, win 245, length 397 15:01:35.863823 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 3656, win 2050, length 0 If I check the Kamailio log file : Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) exec: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=613 a=26 n=sl_send_reply Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/msg_translator.c:162]: check_via_address(): (52.114.76.76, 52.114.76.76, 0) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} sl [sl_funcs.c:500]: sl_run_callbacks(): execute callback for event type 1 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} sl [sl_funcs.c:500]: sl_run_callbacks(): execute callback for event type 1 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace.c:1364]: trace_sl_onreply_out(): trace off... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2225]: tcpconn_send_put(): send from reader (17837 (25)), reusing fd Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2460]: tcpconn_do_send(): sending... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2494]: tcpconn_do_send(): after real write: c= 0x7f81740edb30 n=397 fd=8 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2495]: tcpconn_do_send(): buf= Jan 24 15:01:35 mysbc kamailio[17788]: #027#003#003#001...M...t.[.._..gu..-3.8D#027#022..J.>..a..H.AË.. ..)om.Oh.O.1.A..'.Fa...c.9..M}...#004O..#004 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:498]: pipport2su(): the port string is 5061 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:498]: pipport2su(): the port string is 5061 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/proxy.c:264]: mk_proxy(): doing DNS lookup... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:302]: trace_send_hep2_duplicate(): setting up the socket_info Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) exec: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=614 a=2 n=exit Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/receive.c:353]: receive_msg(): request-route executed in: 1018 usec Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/receive.c:457]: receive_msg(): cleaning up So, it seems that Kamailio is reusing the existing tls connection and then sends my 200 OK to the port 3008 and not to the port 5061 as asked in the contact header. Is my understanding ok? Is that a normal behavior? Thanks for your help. Best regards, Cyrille

1 1

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

24 Jan '20

Hello, Kamailio SIP Server v5.2.6 stable release is out. This is a maintenance release of the previous stable branch, 5.2, that includes fixes since the release of v5.2.5. There are no changes necessary to database schema or configuration language structure if you are upgrade from an earlier installation of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.6. For more details about version 5.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/01/kamailio-v5-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> Kamailio Merchandising - https://skalatan.de/merchandising/

1 0

Implicit branch creation and private variables questions
by George Diamantopoulos 24 Jan '20

24 Jan '20

Hello all, I'm trying to accomplish the following scenario, and some questions have been raised during testing: I'd like to be able to fork serially to a number of downstream destinations in case of failure, but also try several hosts which are available per destination network before failing over to the next one. To that end, I'm using something similar to the following: request_route { ... $vn(some_var) = "some_value"; $vn(some_other_var) = "some_other_value"; ... $avp(provider_order) = "last_provider"; $avp(provider_order) = "first_provider"; ... t_set_fr(120000, 2000); route(PROVIDER_SELECTION); ... } route[PROVIDER_SELECTION] { if ( is_avp_set("$avp(provider_order)") ) { t_on_failure("PROVIDER_SERIAL"); if ( $avp(provider_priority) == "first_provider" )

4 6

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

5 8

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/

1 0

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 >> >

1 0

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

2 1

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

1 0

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

1 0

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

2 1

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

3 4

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

2 1

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

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