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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

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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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