- sr-users - kamailio.org

TAS recommendation
by BALL SUN 23 Jul '20

23 Jul '20

Hi We recently setup Kamailio as the IMS core in our test lab, and we would like to test scenarios with TAS, is there any recommendation? Does Kamailio able to setup as a SIP AS? if yes, any online instruction on how we can make it works? Thanks RBK

1 0

Dispatcher does not go to 2nd server
by Aristeidis Tsitras 22 Jul '20

22 Jul '20

I have a Kamailio and 2 asterisk servers. All users are created in both of the asterisk servers. I am forwarding the registration to asterisk. The problem is that it is always used on only one server from the list. Even if one goes to shutdown, then there is not any registration sent to the available server. Even if *some *of the extensions can be seen registered in both of the asterisk's, if the secondary goes down, then there are no services for the phones. I am attaching the kamailio.cfg. My dispatch list is: 1 sip:192.168.0.100:5080 0 0 maxload=20 2 sip:192.168.0.101:5080 0 0 maxload=20 In both of the asterisk servers i am using sip.conf to create users and s sip trunk for the Kamailio. Nothing special about it. I am looking to find what is wrong with my config and i cannot loadbalance/failover to the asterisk servers.

6 12

Siremis - PassThru Authentication
by Adam - UC-Reseller 21 Jul '20

21 Jul '20

I have recently installed Siremis And have set up a domain pointing to it, how do I configure it so registration requests to this domain are passed thru to the actual PBX Adam

2 1

P-Asserted-Identity in SIP INVITE from Kamailio IMS to PSTN (MGCF)
by BALL SUN 21 Jul '20

21 Jul '20

Hi We have a question. We are preparing a setup to bridge the SIP call from Kamailio IMS to PSTN, but we are puzzling how we can present the P-Asserted-Identity in the SIP INVITE. By default, the PAI is in sip:user@domain, but we believed we should use<tel> format in the PAI when bridge to PSTN. question is 1. should we have two PAI headers?, or 2. should we paste <tel> format in PAI User Part and domain in PAI Host Part?, or 3. Should we only use one PAI header with tel format? In our test setup, we are using FHoSS, and the IMPU sip:user@domain have below entry in "IMPUs from Implicit-Set" - sip:user@domain - tel:number Question is how we can extract the number from the HSS and presented in PAI? - RBK

1 1

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would like tohave: -kamailio (5.3 from deb repo) in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16 LTS). IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the Kamailio setup and users can register on Kamailio following the basic setup. I would like advice on setup and configuration, to integrate Kamailio with Asterisk boxes Some advice please? Sincerely yours, John

2 2

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16. IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3. 0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the Kamailio setup and users can register on Kamailio following the basic setup. I would like advice on setup and configuration, to integrate Kamailio with Asterisk boxes

1 0

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.218 -Asterisk _1. IP: 192.169.1.232.It will serve as media server for IVR, VM, etc -Asterisk-2. IP: 192.168.1.233. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the I would like advice on setup and configuration,

1 0

Kamailio & Freeswitch as media and SBC
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup Kamailio with Freeswitch in a Debian machine as in the http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc, but Kamailio fails to start and gives a ton of errors. Before of the alterations Kamailio and Siremis start correctly and i can easily create users through Siremis. After the changes to include Freeswitch's blocks, it fails to start. All run in the same machine with IP 192.168.1.213. I am attaching kamailio.cfg (named as test1.txt) if someone could spot the error, please. Also i am attaching the kamailio.cfg(Original1.txt) when it was working fine with Siremis.

3 8

SIP Security handling in Kamailio IMS
by BALL SUN 20 Jul '20

20 Jul '20

Hi During the testing with Kamailio IMS, we found that SUBSCRIBE and NOTIFY message flow between UE and P-CSCF is shown below. SUBSCRIBE and its response: 192.168.1.102:9101 (port-c) -> 192.168.2.66:6101 (port-s) NOTIFY: 192.168.2.66:5101 (port-c) -> 192.168.1.102:9100 (port-s) based on the sip security, is it the expected behavior that they are using port-c and port-s? The reason why we asked is because we found that the NOTIFY did not properly propagate to the UE due to different port addresses. Does anyone have this experience? and how we can resolve it? - RBK

2 2

the list with sip compact header names
by Daniel-Constantin Mierla 17 Jul '20

17 Jul '20

Hello, anyone knowing by heart or a place listing all compact header names? I found: - https://www.cs.columbia.edu/sip/compact.html but it says: Last updated 01/10/2008 Maybe newer RFC introduced other compact header names. Or, in case you are aware of some other compact names not listed there, just say it here, as I am working a bit on updating the parser on this scope. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

CommCon 2020 Online - Now presentation about Kamailio
by Daniel-Constantin Mierla 16 Jul '20

16 Jul '20

Hello, being a bit late with this announcement, as it started a couple of minutes ago, but if anyone wants to watch the live presentation about Kamailio by Fred Posner at CommCon 2020 Online, here is the link: - https://www.youtube.com/watch?v=FGZF7dYdRng Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Reset Prometheus counter once
by Marat Gareev 16 Jul '20

16 Jul '20

Hello. Are there any ways to reset the Prometheus counter once? I can do something like this, but counter will be reset on each request loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "prom_counter", "name=mycounter;") > route { > prom_counter_reset("mycounter"); > } Otherwise, I will not receive the counter until a special and rare event occurs. Regards, Marat

3 4

RPC Socket stops answering after a few time.
by David VILLAUME 16 Jul '20

16 Jul '20

Hello, I’m having problem with RPC socket that stops answering after approximatively one or two hours. Version is 5.2.7 My principal host with exact same version and configuration has no problem, only the backup node (Keepalived + DMQ) encounters the problem. I could catch this log about the problem, but didn’t find any reason for this : Jun 18 10:53:33 sbc-kam-partner-b kamailio[83170]: ERROR: ctl [binrpc_run.c:600]: rpc_send(): ERROR: binrprc: rpc_send: send failed My configuration about rpc : loadmodule "jsonrpcs.so" modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") modparam("ctl", "binrpc_max_body_size", 1024) Do you have any idea to troubleshoot this ? Regards, David

2 13

Routing specific number from kamailio to asterisk
by Hendra 834 16 Jul '20

16 Jul '20

Hi, Im new in VoIP, especially Kamailio. I want to connect my kamailio with asterisk by routing the call coming from kamailio to asterisk for specific number, also from asterisk to kamailio. For example, if kamailio client call ext number 0123, the will be routed to asterisk, if kamailio client call 0233 the call will not be routed to external proxy. And also how can I receive call form asterisk to kamailio ? Is there any real example, ? because I dont understand what to change in documentation example. Thanks

2 1

Is it possible to load both dialog and ims_dialog modules in the same configuration?
by Sumesh Soman 16 Jul '20

16 Jul '20

Hi, Is it possible to load and use both dialog and ims_dialog modules in the same server ? Even though ims_dialog docs says ims_dialog extends dialog module, I do see that some of the dialog module features are not present in ims_dialog. Example, wait_ack , track_cseq_updates are not present in ims_dialog. I have requirements to use ims_dialog mainly for ims_charging module and dialog module with wait_ack and track_cseq_updates parameters are enabled. -- Regards, Sumesh S.

1 0

5.4 compiler warning
by Juha Heinanen 14 Jul '20

14 Jul '20

Got this when building 5.4; CC (gcc) [sip-proxy] core/dset.o core/dset.c: In function 'uri_trim_rcv_alias': core/dset.c:1042:6: warning: variable 'proto' set but not used [-Wunused-but-set-variable] int proto; ^~~~~ -- Juha

2 2

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 13 Jul '20

13 Jul '20

Hello, Karsten, Thanks for the link. I will definitely examine it.

1 0

No of registartions
by Pavithra Mohanraja 13 Jul '20

13 Jul '20

Hi, Is there any way to query number of registrations happened in kamailio IMS. Kindly reply . Thanks, Pavithra

2 1

subsequent ds_select_dst with different groups
by David VILLAUME 13 Jul '20

13 Jul '20

Hello, Is it possible to use the following logic : Route() ds_select_dst (1,2) failure_route(1) ds_next_dst(first_dispatcher_ secondgateway) failure_route(newdispatcher_group) ds_select_dst (2,2) failure_route(secondispatcher_secondgateway) ds_next_dst() Thanks in advance. Regards, David

4 5

Compact SIP
by Abdoul Osséni 13 Jul '20

13 Jul '20

Hello, I use Kamailio and Asterisk. SIP Compact header is activated by ASTERISK ( https://www.cs.columbia.edu/sip/compact.html). I am trying to disable the SIP Compact header on KAMAILIO. Is it possible? Regards Abdoul OSSENI

3 4

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 12 Jul '20

12 Jul '20

I'm sorry. I mean *xhttp_prom.c* file instead of *xhttp_mod.c*. >

1 0

xHTTP_PROM with older Kamailio
by Marat Gareev 11 Jul '20

11 Jul '20

Hello, I have a customized Kamailio (kamailio 5.1.1 (x86_64/linux)), xHTTP_PROM <https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html> was released in the 5.3.0 <https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/> version. I'm trying to import the latest version of xhttp_prom to older Kamailio, but something goes wrong. This is simple config for test: log_stderror=yes > listen=5060 > debug=2 > mpath="/home/devel/build_dir/build/lib64/kamailio/modules" > loadmodule "tm.so" > loadmodule "sl.so" > loadmodule "xlog.so" > loadmodule "pv.so" > disable_tcp=no > tcp_accept_no_cl=yes > loadmodule "xhttp.so" > loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "xhttp_prom_stats", "all") > route { > sl_send_reply("200", "OK"); > exit; > } > event_route[xhttp:request] { > if (prom_check_uri()) { > prom_dispatch(); > } else { > xhttp_reply("404", "Not Found", "text/plain", "Wrong URL $hu\n"); > } > } then I send HTTP request $ curl http://localhost:5060/metrics but get error: > 6(110968) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 11 > 0(110962) ALERT: <core> [main.c:746]: handle_sigs(): child process 110967 > exited by a signal 11 > 0(110962) ALERT: <core> [main.c:749]: handle_sigs(): core was generated > 0(110962) INFO: <core> [main.c:771]: handle_sigs(): terminating due to > SIGCHLD note: for http://localhost:5060/test request I get the expected message "Wrong URL /test". Backtrace for killed process > (gdb) c > Continuing. > Program received signal SIGSEGV, Segmentation fault. > 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > 224 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) > (gdb) bt > #0 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > #1 0x00007fefff3ff0e2 in futex_get (lock=0x0) at > ../../core/mem/../futexlock.h:99 > #2 0x00007fefff4130bf in prom_metric_list_print (ctx=0x7fefff628da0 > <ctx>) at prom_metric.c:1319 > #3 0x00007fefff416468 in prom_stats_get (ctx=0x7fefff628da0 <ctx>, > stat=0x7fefff6281f0 <xhttp_prom_stats>) at prom.c:178 > #4 0x00007fefff3d4d10 in prom_send (ctx=0x7fefff628da0 <ctx>) at > xhttp_prom.c:361 > #5 0x00007fefff3d5faf in ki_xhttp_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:418 > #6 0x00007fefff3d5feb in w_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:428 > #7 0x0000000000461dbe in do_action (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1067 > #8 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1565 > #9 0x0000000000461d2d in do_action (h=0x7ffecb730e00, a=0x7ff0007ca628, > msg=0x7ffecb730ee0) at core/action.c:1058 > #10 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c76f8, > msg=0x7ffecb730ee0) at core/action.c:1565 > #11 0x00007fefff62f649 in xhttp_process_request (orig_msg=0x7ff0007ebbd8, > new_buf=0x7ff0007cc280 "GET /metrics HTTP/1.1\r\nVia: SIP/2.0/TCP > 127.0.0.1:37000\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", new_len=119) at xhttp_mod.c:298 > #12 0x00007fefff630d78 in xhttp_handler (msg=0x7ff0007ebbd8) at > xhttp_mod.c:385 > #13 0x0000000000520141 in nonsip_msg_run_hooks (msg=0x7ff0007ebbd8) at > core/nonsip_hooks.c:111 > #14 0x000000000058cef1 in receive_msg (buf=0x7feffee65bb0 "GET /metrics > HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: localhost:5060\r\nAccept: > */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0) > at core/receive.c:216 > #15 0x000000000061233f in receive_tcp_msg (tcpbuf=0x7feffee65bb0 "GET > /metrics HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0, > con=0x7feffee658b8) at core/tcp_read.c:1379 > #16 0x000000000061479e in tcp_read_req (con=0x7feffee658b8, > bytes_read=0x7ffecb731c7c, read_flags=0x7ffecb731c78) at > core/tcp_read.c:1611 > #17 0x00000000006174e9 in handle_io (fm=0x7ff0007d9720, events=1, idx=-1) > at core/tcp_read.c:1785 > #18 0x00000000006069b4 in io_wait_loop_epoll (h=0xaa76c0 <io_w>, t=2, > repeat=0) at core/io_wait.h:1065 > #19 0x00000000006194ff in tcp_receive_loop (unix_sock=14) at > core/tcp_read.c:1955 > #20 0x00000000004f1a43 in tcp_init_children () at core/tcp_main.c:4853 > #21 0x0000000000423f99 in main_loop () at main.c:1716 > #22 0x000000000042a5ee in main (argc=9, argv=0x7ffecb7323b8) at main.c:2646 > (gdb) > (gdb) p 'prom_metric.c'::prom_lock > $4 = (gen_lock_t *) 0x0 If I understood correctly, the problem is in an uninitialized variable static gen_lock_t *prom_lock = NULL; /* Lock to protect Prometheus metrics. */ So, is it possible to import xHTTP_PROM to older Kamailio? I will be glad for any advice. Thanks.

1 0

loose_route failing with topos ?
by Mark Boyce 10 Jul '20

10 Jul '20

Hi all Just enabled topos with redis backend (topos/topos_redis/ndb_redis) The interesting bits of the calls are going from kamailio (5.3.5) to Asterisk with topos enabled for that leg All seems to be ok until the far end (Asterisk) sends a BYE. At which point kamailo passes message to WITHINDLG, in loose_route comes up false (-1) and the BYE gets missed, falling through to the “404 Not here” at the end of WITHINDLG. I’m guessing that loose_route should be able to “see inside” topos, or should I be throwing loose_route away when using topos? Happy to upgrade to 5.4 and do diags if it appears to be a bug. Best regards Mark -- Mark Boyce Dark Origins Ltd

3 4

rtcp-mux is missing in the OK for WebRTC-Calls
by Benjamin Flügel | vio:networks 10 Jul '20

10 Jul '20

Hey guys, I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS using WebRTC. So far it works great on Firefox but have a specific problem with chrome, when I want to make call from the softphone to another extension. After anwsering the call Chrome/the softphone sends a BYE immediately, because this line "a=rtcp-mux" is missing in the OK. The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible for the pbx-features. Those are my rtpengine Flags for the Invite: rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP And those are the flags for the response, in this case the OK: rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal direction=external loop-protect It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the response. Can you help me get it to work? Here is the SIP-Dialog. The call is from extension 201 to the extension 2. INVITE: INVITE sip:2@mydomain SIP/2.0 Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;branch=z9hG4bK5104892 To: <sip:2@mydomain.com> From: <sip:201@mydomain.com>;tag=aqplr71k05 CSeq: 2 INVITE Call-ID: sde09v49f8gqtt59oddn Max-Forwards: 70 Proxy-Authorization: Digest algorithm=MD5, username="201", realm="mydomain.com", nonce="sfdsdfdsfsdfdsfsdfaseqww", uri="sip:2@mydomain.com", response="81rewtrega23423r" Contact: <sip:kf6iduon@g4n0lpfirgpv.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIPJS Content-Type: application/sdp Content-Length: 1916 v=0 o=- 5826889093965459811 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO m=audio 54274 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 mycomputerIP a=rtcp:9 IN IP4 0.0.0.0 a=candidate:1577908739 1 udp 2113937151 efd6d297-d186-4c07-87d8-933e5846a82d.local 54274 typ host generation 0 network-cost 999 a=candidate:842163049 1 udp 1677729535 mycomputerIP 54274 typ srflx raddr 0.0.0.0 rport 0 generation 0 network-cost 999 a=ice-ufrag:Ktxl a=ice-pwd:x5xDUb41GeOXAcHNQlra4yUN a=ice-options:trickle a=fingerprint:sha-256 2C:7C:32:AF:34:A3:9D:AE:C7:FD:92:68:DD:D8:AB:82:DB:F0:32:51:14:97:20:60:66:5C:2F:CF:B7:98:B8:A8 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2376769177 cname:RgPkYyxDrEJTpmO2 a=ssrc:2376769177 msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=ssrc:2376769177 mslabel:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO a=ssrc:2376769177 label:dc187be3-4cd0-4129-98e4-f804cd8d2c94 RINGING: SIP/2.0 180 Ringing Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internalIP;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:externalIP;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Length: 0 OK: SIP/2.0 200 OK Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internal-ip;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:external-ip;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Type: application/sdp Content-Length: 806 v=0 o=root 882840402 882840402 IN IP4 externalIP s=Asterisk PBX 11.22.0 c=IN IP4 externalIP t=0 0 a=rtpengine:4d633e3022f5 m=audio 16416 RTP/SAVPF 111 9 8 0 126 a=maxptime:60 a=mid:0 a=rtpmap:111 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0 a=fmtp:126 0-16 a=sendrecv a=rtcp:16417 a=setup:active a=fingerprint:sha-1 B1:67:4B:B8:47:89:E8:49:CD:DD:F8:FF:41:5C:83:72:D9:DE:4D:45 a=ptime:20 a=ice-ufrag:WBZN8m7c a=ice-pwd:mKVR6Jv0G0pvrTv7OfEm2wPW9Q a=ice-options:trickle a=candidate:3kddB3zBUV2n2jt0 1 UDP 2130706431 externalIP 16416 typ host a=candidate:3kddB3zBUV2n2jt0 2 UDP 2130706430 externalIP 16417 typ host a=end-of-candidates I hope, you can help me with this. If you have further question, I will try to answer them as best I can. Greetings, Ben

2 1

Development is open for future release of v5.5.x series
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 was created, therefore the master branch is open for adding new features, to be part of future release series v5.5.x. Any bug fix committed to master that applies to 5.4.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.4 or 5.3. Expect that v5.4.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.5.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime in the first part of 2021. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Branch 5.4 has been created
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 has been created, to be used for releasing v5.4.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.4 cd kamailio-5.34 git checkout -b 5.4 origin/5.4 Pushing commits in this branch: git push origin 5.4:5.4 Note that 5.4 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.4 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.4 will be out, respectively Kamailio v5.4.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Drop local generated requests
by Patrick Wakano 10 Jul '20

10 Jul '20

Hello list, Hope you all doing fine! I have a case where I want to drop a local generated request but it looks like you can't drop the request from the tm:local-request route. I found this ticket https://github.com/kamailio/kamailio/issues/980, but I could not find somewhere in the docs or emails definitely saying that drop() is not supported from tm:local-request.... And in case we definitely can't drop the request in there, any ideas on how to avoid a local generated request of going out? I am specifically interested in dropping Registers from the uacreg module under certain situations, but I don't want to disable it or delete it from the DB.... Thank you, Kind regards, Patrick Wakano

2 4

Help on dumping running config in Kamailio
by BALL SUN 09 Jul '20

09 Jul '20

Hi I need an urgent help, I accidentally pipe the output to the kamailio config, and now I lost the copy of my latest setting, is there a chance that I can dump the running config back to file? Thanks PLEASE HELP RBK

2 1

Branching 5.4 in git repository
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, so far it looks pretty calm during the testing phase, some static analysis didn't reveal any new major issues (well, this is done also during development phase). Therefore, if nothing else is proposed or shows up, I am considering to create the git branch 5.4 next week on Thursday -- this branch will hold the code for 5.4.x release series. After branching 5.4, the master branch will be open again for new features. The 5.4.0 enters the release candidate phase, likely to have the first release out in 1-2 weeks after branching. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

Maintenance work on kamailio.org server - Wed, July 8, 2020
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, on Wednesday, July 8, 2020, there will be some work on Kamailio project infrastructure. Therefore there could be short intervals of downtime for online resources like the web server, wiki, documentation and downloads portals, mailing lists or the GIT repository mirror. Once the maintenance work is finished, we will post an update. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

30X redirect in interconnects - better alternative?
by Gerry | Rigatta.com 08 Jul '20

08 Jul '20

Hi, I would like to use Kamailio for load balancing incoming carrier traffic. We do currently IP authentication and call logic in Yate boxes. Ideally I would like to distribute calls with 30X redirects with the Kamailio dispatcher so that IP authentication and all logic can stay in the Yate boxes. However I have doubts that 30X redirects are generally accepted in interconnects. What is your experience with this? What is the possible alternative to redirects if one wants to keep IP authentication and call logic in the boxes behind the Kamailio SIP router? E.g. how can one reliably check the carrier source IPs behind Kamailio? Custom headers injected by Kamailio? Of cause I can check source IPs with a database lookup in Kamailio but I try to avoid that as this makes the setup much more complicated and error prone. Thank you for your ideas. Gerry

3 2

Re: [SR-Users] sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Update: Disabling the topoh module on the proxy which produces the error seems to stop the failure from manifesting. I'll try using topos_redis instead, but should this be treated as a bug? BR, George On Wed, 8 Jul 2020 at 12:37, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello again, > > Indeed $mb seems to contain garbage: > > SCRIPT_MB: ACK <BC><EA><8F> SIP/2.0#015#012Via: SIP/2.0/UDP > 172.30.154.189;TH=dcv;branch=z9hG4bK629b.6af9302cd78dc58dffe817e60124f4ed.0#015#012Route: > <sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@172.30.155.1 > ;lr;received=sip:2.2.2.2:32768 > ;ob;r2=on>,<sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@3.3.3.3 > ;lr;received=sip:2.2.2.2:32768;ob;r2=on>#015#012Max-Forwards: > 68#015#012From: "Anonymous" <sip:unknown@voip.domain.com>;tag=as4bc9e324#015#012To: > <sip:voip-test-user-02@voip.domain.com>;tag=jw7z5s0zvc#015#012Call-ID: > 0138c6370346d1dd7f1b47f604b01f92(a)voip.domain.com#015#012CSeq > <http://0138c6370346d1dd7f1b47f604b01f92@voip.domain.com#015%23012CSeq>: > 102 ACK#015#012Content-Length: 0#015#012TH: dch#015#012#015#012 > > How can this be possible? Capturing traffic on wire shows the RURI I > pasted in my original message and there are no script operations on the > RURI before sanity_check() (message buffer above is printed just before > sanity_check() is run in REQINIT). > > BR, > George > > On Wed, 8 Jul 2020 at 11:18, George Diamantopoulos <georgediam(a)gmail.com> > wrote: > >> I'll post the message buffer ASAP, but in the meantime I don't see how >> config operations could affect the RURI. Here's everything that's happening >> until the sanity check involved: >> >> request_route { >> >> if(is_method("KDMQ")) { >> dmq_handle_message(); >> } >> >> # no connect for sending replies >> set_reply_no_connect(); >> >> if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { >> # silent drop for scanners - uncomment next line if want to reply >> # sl_send_reply("200", "OK"); >> exit; >> } >> >> if (!mf_process_maxfwd_header("10")) { >> force_rport(); >> sl_send_reply("483","Too Many Hops"); >> exit; >> } >> >> # OPTIONS and NOTIFYs directed to myself >> if(is_method("OPTIONS|NOTIFY") && uri==myself && $rU==$null) { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> # All keep-alive methods regardless of destination >> if ( $hdr(Event) == "keep-alive") { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> if(!sanity_check("17895", "7")) { >> xlog("Malformed SIP request from $si:$sp\n"); >> exit; >> } >> >> BR, >> George >> >> On Wed, 8 Jul 2020 at 10:58, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> check your config operations, because the R-URI seems to be the next >>> string (without quotes): "<BC><EA><8F>" >>> >>> You can try to print $mb in such case to see the entire SIP message >>> buffer. >>> >>> Cheers, >>> Daniel >>> On 08.07.20 09:48, George Diamantopoulos wrote: >>> >>> Hello Daniel, >>> >>> Thanks for the reply. Indeed there is, not sure how I managed to miss >>> that. And it wasn't about the schema after all: >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1254]: parse_uri(): uri too short: <<BC><EA><8F>> >>> (3) >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1328]: parse_sip_msg_uri(): bad uri <<BC><EA><8F>> >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: WARNING: >>> {1 <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:282]: check_ruri_scheme(): failed to parse request uri >>> [<BC><EA><8F>] >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: ERROR: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <script>: Malformed >>> SIP request from 172.30.154.189:5060 >>> >>> Still, not sure what the problem is though... >>> >>> BR, >>> George >>> >>> On Wed, 8 Jul 2020 at 09:30, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> when the ruri scheme check fails, there should be another debug message >>>> saying that. Have you pasted all log messages for the failure case? >>>> >>>> Cheers, >>>> Daniel >>>> On 07.07.20 22:23, George Diamantopoulos wrote: >>>> >>>> Sorry, I realised I copy pasted wrong log messages for Call 1. Here's >>>> the relevant part showing success for call 1 in contrast with Call 2: >>>> >>>> grep 2a859fcc4e1c8f840191a81d7c16e76d kamailio.log | egrep >>>> 'check_ruri_scheme|w_sanity_check' | grep ACK >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 >>>> >>>> On Tue, 7 Jul 2020 at 21:34, George Diamantopoulos < >>>> georgediam(a)gmail.com> wrote: >>>> >>>>> Hello all, >>>>> >>>>> I'm not 100% sure this is the only culprit in an issue I'm >>>>> investigating, but superficially it appears that RURI scheme sanity module >>>>> checks from the default config (flags 17895 in REQINIT) fail if the >>>>> RURI in an ACK following a 487 includes parameters. Example from two calls >>>>> from a kamailio instance acting as registrar/usrloc server, INVITE RURIs >>>>> are after usrloc lookup: >>>>> >>>>> Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> These INVITEs produce no complaints. Later, the same registrar >>>>> produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses >>>>> from the next proxy in the path following a CANCEL: >>>>> >>>>> Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> The next proxy (which produced/relayed the 487) processes the ACK for >>>>> Call 1 successfully, but sanity_check at the proxy drops the request for >>>>> Call 2 with: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>>>> >>>>> whereas Call 1 seems OK: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:305]: check_required_headers(): check_required_headers entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:313]: check_required_headers(): check_required_headers passed >>>>> >>>>> Could this be a bug in sanity module? Is there anything one can do in >>>>> config which could result in illegal ACKs being produced for hop-by-hop >>>>> transactions? schema appears to be sip: in both cases... >>>>> >>>>> Thank you. Best regards, >>>>> George >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Funding: https://www.paypal.me/dcmierla >>>> >>>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>>

2 2

Setting up Kamailio as Teams SBC
by Роман С. 08 Jul '20

08 Jul '20

Hello. I'm trying to set up Kamailio as SBC for Teams using https://skalatan.de/en/blog/kamailio-sbc-teams. Setup is completely default (except things mentioned at article), but I use wildcard certificate for TLS. Well, I can't even pass dispatcher: kamcmd dispatcher.list | egrep "RI|FLAG" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 3 URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 2 URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 1 Where do I start to dig?

3 6

sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Hello all, I'm not 100% sure this is the only culprit in an issue I'm investigating, but superficially it appears that RURI scheme sanity module checks from the default config (flags 17895 in REQINIT) fail if the RURI in an ACK following a 487 includes parameters. Example from two calls from a kamailio instance acting as registrar/usrloc server, INVITE RURIs are after usrloc lookup: Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 These INVITEs produce no complaints. Later, the same registrar produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses from the next proxy in the path following a CANCEL: Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 The next proxy (which produced/relayed the 487) processes the ACK for Call 1 successfully, but sanity_check at the proxy drops the request for Call 2 with: DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 whereas Call 1 seems OK: DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:305]: check_required_headers(): check_required_headers entered DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:313]: check_required_headers(): check_required_headers passed Could this be a bug in sanity module? Is there anything one can do in config which could result in illegal ACKs being produced for hop-by-hop transactions? schema appears to be sip: in both cases... Thank you. Best regards, George

2 5

Some Functions not defined in Documentation of Module ims_registrar_pcscf
by Hamid Hashmi 08 Jul '20

08 Jul '20

I have been using a sample code of PCSCF<https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/pcscf/ro…>, where a function "pcscf_assert_identity" is used but it's not defined in module ims_registrar_pcscf documentation. Can you please explain its functionality. The same way other functions are not defined in the documentation. pcscf_assert_called_identity reginfo_handle_notify pcscf_unregister Regards Hamid R. Hashmi

1 0

Kamailio 5.3.4 - Segfault on http_async_client
by José Seabra 07 Jul '20

07 Jul '20

Hello there, Kamailio is terminating with signal 11 and is generating a coredump file. I opened the core file generated by it and it seems to be related to the module http_async_client. In the following link you can find the output of core dump backtrace full: https://pastebin.com/rapPZy9N Below is the http_async_client config that I have: modparam("http_async_client", "workers", 2) modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "hash_size", 2048) modparam("http_async_client", "curl_verbose", 0) modparam("http_async_client", "memory_manager", "shm") If the information provided is not enough to troubleshoot this issue let me know. Many thanks Best Regards -- Cumprimentos José Seabra

1 1

Kamailio Buildroot
by Benjamin Marty 07 Jul '20

07 Jul '20

Hello I would like to ask if someone already tried to build Kamailio for the Embedded Distribution Buildroot. I tried it myself but it looks like Kamailio is not that easy to cross compile. Thanks Benjamin

3 3

siptrace not sending data to hepic
by Andrew Chen 07 Jul '20

07 Jul '20

Hi all, I've been assigned a task to investigate why hepic doesn't generate MOS scores ever since we changed the kamailio interface MTU from 9001 -> 1500. During my investigation, this is what I have found out so far: - Putting MTU back to 9001 removes this condition where no MOS was generated. - I did a tcpdump of port 9063 which is the port used to send data to Hepic capture server and saw it reports packets are fragmented: 3:48:55.895505 IP (tos 0x68, ttl 64, id 38049, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP 13:48:55.896451 IP (tos 0x68, ttl 64, id 38050, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP - Hepic support also caused by using 1500 MTU on the interface. - I tried using the following core parameters to see if they could remedy this issue but didn't seem to work: ## MTU related core settings udp4_raw=-1 udp4_raw_mtu=1500 I can't seem to find any parameters in the siptrace module that controls fragmentation of the packets to Hepic. Any suggestions on how else to troubleshoot this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Keep one registration per device(src_ip+port)
by Zhan Bazarov 07 Jul '20

07 Jul '20

Hello. I'd like to keep one registration per device in usrloc table, for other cases I want to store and maintain one contact per AoR. Does kamailio support it out of the box? For second case save("location", "0x01"); working fine. But how about first case? for example, if one of device located at home, and second located at office. - there are two different network and NAT - how I can make both devices ringing when INVITE is sent? If I do classic multiAor scenario - it can lead to issues caused by multiple contacts in 200OK response for a single device and devices are confusing sometimes. So, what I did now: if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); add_contact_alias(); } [REGISTRAR] { if (nat_uac_test(64)) { if (!save("location", "0x04")) { sl_reply_error(); } } else { if (!save("location")) { sl_reply_error(); } } } modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "xavp_contact", "usrlocattrs") modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "expires_type", 1) modparam("usrloc", "matching_mode", 1) modparam("registrar", "method_filtering", 1) modparam("registrar", "max_expires", 3530) modparam("registrar", "min_expires", 1800) modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("registrar", "gruu_enabled", 0) modparam("registrar", "use_path", 1) modparam("registrar", "path_check_local", 1) modparam("registrar", "path_mode", 1) modparam("path", "use_received", 1) How do u think, maybe modparam("registrar", "append_branches", 0) will make some help? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Nathelper ping from dedicated kamailio instance
by Angelo Pellegrinon 07 Jul '20

07 Jul '20

Hi! I am new to kamailio so please forgive me if I write some errors. I am configuring some kamailio instances in different environments to handle big traffic and I have a problem with usrloc + db + nathelper. What have I done: 1. Configured a kamailio instance to handle sip registers with usrloc to database with database mode 2 (memory + db), disabled nathelper pings. This is ok and performs well; 2. Configured a kamailio instance to handle nathelper ping. Same usrloc + db + nathelper config from 1 except that I use db mode 3 (only db) and that I enabled natping_interval . My problem is that it seems that the second kamailio instance does not send pings. The second one connects correctly to db (if I lookup from kamcmd a user registered from the first kamailio I can find it) but it seems that does not read all locations. Can you help me? Many many thanks Angelo

1 0

rtpengine with mutiple IPs on the same network.
by David Villasmil 07 Jul '20

07 Jul '20

Hello guys, I have rtpengine running on a box with ips like this: external: 172.10.0.10 (advertise 10.10.10.1) internal: 172.10.0.11 Both Ips are on the same network. When i do rtpengine_manage i set "direction=external direction=internal", and the SDP offer is correct. The problem I'm having is that when rtp tried to send to the internal network, the OS (i believe) selects the first interface (external) ip 172.10.0.10 to reach the internal network. This I suppose is not a great deal, but I would like to separate them and only use the internal for internal proxying. Help is appreciated! Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

Retrieving cert details from tls peer
by Mark Boyce 07 Jul '20

07 Jul '20

Hi all Been trying to grab the TLS cert details from incoming connections, but failing :-( So with lines just before AUTH is called like this; if (proto == TLS) { xlog("L_INFO", "TLSDUMP $ci peer_subject : $tls_peer_subject\n"); Gets met with a log line line this; INFO: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 1.2.3.4:11797 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 INFO: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: 5.6.7.8:5061 INFO: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate ... INFO: tls [tls_select.c:168]: get_cert(): Unable to retrieve peer TLS certificate from SSL structure This is with verify_certificate and require_certificate set to no in tls.cfg If I try and set the following in tls.cfg [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = yes I see in the logs; INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=22 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/tls-certs/cert.pem' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/tls-certs/privkey.pem' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 NOTICE: tls [tls_domain.c:1095]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... INFO: tls [tls_domain.c:692]: set_verification(): TLSs<default>: Client MUST present valid certificate INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 INFO: tls [tls_domain.c:692]: set_verification(): TLSc<default>: Server MUST present valid certificate ... ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Which looks like verification is being enabled when I add require? Would someone be kind enough to point out what I am missing please? (Assuming it’s not a bug :-) Thanks Mark -- Mark Boyce Dark Origins Ltd

2 8

About routing calls
by ahmed moghazy 07 Jul '20

07 Jul '20

Excuse me i already made integration for kamailio with asterisk but i want to make sip calls through kamailio and media fuctions through asterisk , i want kamailio to make calls away from asterisk , i just wanted asterisk to use its dahdi module .’ Thank you.

2 1

Kamailio Push Notification for iOS13
by william anderson 07 Jul '20

07 Jul '20

Hello, Yesterday, there was a new release of Linphone IOS app, but i need help from some kamailio expert how to add PUSH notification to the application to receive incoming calls when the phone is not in use (wake up required) from a kamailio server. I have read on internet old tutorials (tsilo module) but most of them are incomplete or do not explain in an easy way how to do it (for an average kamailio user). Anyone could do an easy to understand tutorial ? Regards, William Anderson

2 1

Looking for a VoIP & Kamailio Dev-Ops
by Henning Westerholt 06 Jul '20

06 Jul '20

Dev-Ops VoIP & Kamailio GILAWA is looking for a new colleague to join our team! Tasks: Deploying, developing, and integrating our software based on Kamailio Testing and validation of our customer solutions Solve complex customer problems in the VoIP field Fix Bugs and create extensions in the Kamailio project Requirements: Good background in VoIP telephony and Kamailio Stack Experience with signalling standards (SIP, RTP etc..) and troubleshooting Knowledge and experience in one of the following languages: C/C++, Bash, PHP Experience with Debian or RedHat/CentOS Linux, knowledge of Ansible would be a plus Fluent in English Based in European or neighbouring time zone (e.g. EMEA area) Self-organized and strong communication skills Given the current situation this is a remote position. If you are interested, please send me a (direct) e-mail with an CV or Linkedin link, please include your availability and salary expectations as well. Best regards, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

3 3

Kamailio 5.3.5, usrloc: Error inserting location (integer out of range)
by Stefan 03 Jul '20

03 Jul '20

Hi Maillist Kamailio-Version: 5.3.5 built from source with postgres-db. I get an Error if I try to add a static location with kamctl, see my example: kamctl ul add 1234 sip:1234@10.10.10.10:5060 NULL NULL Log output: Jul 1 09:28:20 kamailio-srv02 /sbin/kamailio[3892]: ERROR: db_postgres [km_dbase.c:266]: db_postgres_submit_query(): 0x7fbe5398c148 PQsendQuery Error: ERROR: integer out of range#012 Query: insert into location (username,contact,expires,q,callid,cseq,flags,cflags,user_agent,path,methods,last_modified,ruid,reg_id,server_id,connection_id,keepalive,partition ) values ('1234','sip:1234@10.10.10.10:5060','1970-01-01 00:00:00',1.000000 ,'dfjrewr12386fd6-343(a)kamailio.rpc',1,0,0,'kamailio SIP Router - RPC Server','NULL',2147483648,'1970-01-01 00:00:00','ulcx-5ef99724-f37-1',0,0,0,0,22) truncated output of kamctl ul show: "AoRs": [{ "Info": { "AoR": "1234", "HashID": 924651761, "Contacts": [{ "Contact": { "Address": "sip:1234@10.10.10.10:5060", "Expires": "permanent", "Q": 1, "Call-ID": "dfjrewr12386fd6-343(a)kamailio.rpc", "CSeq": 1, "User-Agent": "kamailio SIP Router - RPC Server", "Received": "[not set]", "Path": "NULL", "State": "CS_NEW", "Flags": 0, "CFlags": 0, "Socket": "[not set]", "Methods": -2147483648, "Ruid": "ulcx-5ef99724-f37-1", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": 0, "Keepalive": 0, "Last-Keepalive": 0, "Last-Modified": 0 } }] } }, { So the log tells me: integer out of range. I think it's this one: "Methods": -2147483648, This let's me thinking that this is a bug. Someone else? Regards Stefan

2 8

403 Forbidden error You must register first with a S-CSCF
by Pavithra Mohanraja 03 Jul '20

03 Jul '20

Hi, I am facing below error when i try to make a call using pjsip .. Registration is successful. I am following the open 5gs VolTE setup. I have not changed any configurations in any of the config files. But I am facing below issue .. Can you please help me what could be the reason. *reason=403 (Forbidden - You must register first with a S-CSCF)* Could anyone please help me .. I am ready to provide anyother info if required . Thanks, Pavithra

1 0

Error in SCSCF
by Pavithra Mohanraja 02 Jul '20

02 Jul '20

Hi , I am facing below error in s-cscf when i send a BYE from alice to bob in kamailio IMS. The problem is in only call termination .Kindly help . Error log in scscf : Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7f4746f89148 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 11==11 && [10.244.0.57] == [10.244.0.54] Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:637]: grep_sock_info(): checking if port 4080 (advertise 0) matches port 5060 Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:5060) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0: scscf.sip.example.com:4080) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/forward.c:423]: check_self(): *host (0:10.244.0.57:5060 <http://10.244.0.57:5060>) != me* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: sip:bob at sip.example.com Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *Domain not served 403* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *BYE *( sip:alice@sip.example.com (10.244.0.52:4060) to sip:bob@sip.example.com, -ARSjM.TTZgxhzwDjKRN0DtENWD4bKl0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7f4746dd7010, 30) called from core: core/ut.c: as_asciiz(161) I am not getting what it checks .. 10.244.0.57 -> UE bob 10.244.0.52 -> pcscf 10.244.0.53 -> icscf 10.244.0.54 -> scscf 10.244.0.56 -> UE alice Kindly help Thanks, Pavithra

2 4

Sql_result_free() not cleaning properly
by Duarte Rocha 01 Jul '20

01 Jul '20

Greetings, I'm trying to clean a SQL result using "sql_result_free()" but i'm not having success. Can you please verify if i'm doing anything wrong here? This is my code : xerr("Pre clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); sql_result_free("Cli_PICKER_Default_Search_Result"); xerr("Pos clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); The result is the following : ERROR: <script>:Pre clean [0] = Iterator 0 = 111111111111 ERROR: <script>:Pos clean [0] = Iterator 0 = 111111111111 Am i using sql_result_free correctly or doing something wrong? Best Regards,

4 4

Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start -UPDATE
by Martin W Woscek 30 Jun '20

30 Jun '20

Problem: The mysql db issues with the scscf db include: table version, disappearing impu and/or UE data retained in tables when none are registered Workarround: In order to get over those various mysql issues listed causing the S-CSCF not to come up properly as boot up, 1. create a fresh new scscf database (kamadbctl [drop|create] scscf and import the msql script from the scscf configs) 2. back up the new scscf db 3. modify the init.d startup script for the scscf to restore the fresh db upon each start/restart of the S-CSCF /etc/init.d/kamailio_scscf .. case "$1" in start|debug) mysql -u root --database=scscf < /etc/kamailio/ims_databases/scscf.sql The S-CSCF has been stable and dependable on each boot. _Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? 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 Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

1 0

default config and options for Kamailio 5.4.x
by Daniel-Constantin Mierla 30 Jun '20

30 Jun '20

Hello, with the release of Kamailio 5.4.0 approaching, I opened an item in the tracker to collect suggestions about what should be tuned in the default config file. If you feel like something should be there, add your comment to: * https://github.com/kamailio/kamailio/issues/2381 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Re: [SR-Users] RTP voice failover
by Yufei Tao 30 Jun '20

30 Jun '20

Hi, RTP FO is possible, e.g. with keepalived+rtpengine+Redis as people suggested. Here's the guide for configuring it which I find very useful: https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications A possible way is to have a pair of rtpengine instances in master-slave setup. When master goes down, RTP continues on slave (new master) after a few seconds gap, when the virtual IP(s) migrate over. Another useful link: https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-Pawel.Kuzak-High… Good luck! Yufei <https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications>

1 0

kamailio and rtpengine in docker
by David Villasmil 29 Jun '20

29 Jun '20

Hello guys, I've got a kamailio on a docker and kamailio on another docker on different boxes. The issue i'm having is that when kamailio tries to enable rtpengine it tries pinging it. The ping arrives at the actual container where rtpengine is running, but for some reason i haven't figured out, rtpengine (or the container) is not responding to the ping. I'm writing here in the hopes that someone has already implemented this an come across this and a solved (for sure) I'd appreciate some help. Many thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Valid URL characters in the RURI
by Sam Ware 29 Jun '20

29 Jun '20

I'm using Sanity Checks. Is an underscore (_) not allowed in the host portion of the RURI? I'm on version 5.3. I reviewed the pase_uri function in the core parse_uri.c but I am not super proficient in C. case URI_HOST_P: switch(*p) { check_host_end; default: if(!isalnum(*p) && (*p != '.') && (*p != '-') && !uri_host_char_allowed(*p)) { goto error_bad_host; } } break; I see it calls out periods (.) and dashes (-) but not underscores. I'm wondering why, being underscores are allowed in URLs. -- Sam D Ware

2 1

Changing Debug Level on the fly
by Sam Ware 29 Jun '20

29 Jun '20

I am trying to change the debug level without restarting the service. My research says this is right but I get an error message. $ sudo -u kamailio kamcmd dbg.set_mod_level core 2 error: 500 - cannot store parameter -- Sam D Ware

2 1

SIP OPTIONS between UE and P-CSCF
by BALL SUN 29 Jun '20

29 Jun '20

Hi We have setup UE with Kamailio IMS, and after the registration is successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through IPSEC tunnel. But we found that the SIP OPTIONS (I believed it is a health check) from P-CSCF to UE is not through the IPSEC tunnel. our question is 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside? 2. if it can be configured to run inside the IPSEC tunnel, what configuration can we change to achieve it? - RBK

2 3

How to add parameter ;user=phone in RURI
by Minh Phan 28 Jun '20

28 Jun '20

Hi List, I'm having trouble adding the parameter user=phone in the RURI. I have tried to use the function add_uri_param from siputils module but it doesn't work (see below). Here is the code I use to add user=phone parameter to FROM, TO and RURI uac_replace_from("", "sip:$var(normalized_number)@sip.domain.com ;user=phone"); uac_replace_to("", "sip:$var(normalized_number)@sip.domain.com;user=phone"); $rU = $var(normalized_number); *add_uri_param*("user=phone"); Thanks in advance for any hints, Best regards, Minh Phan

2 1

RTP voice failover
by Evgeniy 27 Jun '20

27 Jun '20

Is that possible to implement RTP voice fail-over? In case RTP node goes down - the call should not be interrupted. This is kinda strange - but client want to have super redundant system. For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 nodes (yes - network traffic will increase) Or maybe is a way to recover UDP connections after a crash and replace node state. Maybe use a Kubernetes tools for that purpose ? Here what i found in mailing-list - https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html Many thanks for any hints to this topic.

6 9

Kamailio IMS with ENUM
by BALL SUN 26 Jun '20

26 Jun '20

Hi Is there any information that can help to setup ENUM query in Kamailio IMS? what we wanted to achieve is to when A in domain A (the local domain) makes an INVITE to B, it will check the domain of B from ENUM backend. if it is not on the local domain, it will route to remote IMS core. Thanks - RBK

2 7

Does kamailio sip domain makes sense if i defined some alias
by PICCORO McKAY Lenz 26 Jun '20

26 Jun '20

in kamctlrc i defined an DOMAIN but later that ip was updated.. so my question: it make sense inclusivelly if already defined more allias in kamailiorc.cfg file? in case of more complicated response,, anybody can explain me about that why? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED
by Martin W Woscek 26 Jun '20

26 Jun '20

Hello, Just wanted to update what fixed our issue. The "contact_user" which is "asterisk" that we were using for the Kamailio context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 16 server was not a valid user in the IMS HSS database (FHoSS implementation). However not being an expert in the IMS, IMHO, the IMS should have kicked out this INVITE as soon as credentials were checked at the front door of the IMS, but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF, which is the correct flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 604 HSS user unknown being sent thru the reverse direction back to Asterisk. Anyway, its always good to end Friday on a high note. Thanks, _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 1:32 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; miconda(a)gmail.com Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Just to clarify what we are seeing, here is a packet from the INVITE from the S-CSCF (6060) to the P-CSCF(4060). I have eliminated the S-CSCF as this packet looks correct. The P-CSCF is not handling the request URI correctly, and simply sends it back to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri updated by the S-CSCF with the local IP address of the UE (192.168.20.1). The P-CSCF should send this INVITE to the UE (703222888(a)192.168.20.1<mailto:703222888@192.168.20.1>) No. Arrival Time Source Destination Source Port Destination Port SIP to tag SIP from tag Method Record-Route Userinfo Request-URI Media Port Protocol Length Request-URI Host Port Owner Username Media Type Media Format Origin-Host Type slice_type SIP from address User Part SIP from address Host Part SIP to address User Part SIP to address Host Part Contact URI User Part Contact URI Host Part Request-URI User Part Request-URI Host Part Request-URI Host Port Info 1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128 192.168.20.128 6060 4060 d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt sip:7032228888@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP 2263 63871 - audio,video ITU-T G.711 PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 7033331111 asterisk.irisims.org 7032228888 ims522.irisims.org asterisk 192.168.20.131 7032228888 192.168.20.1 63871 Request: INVITE sip:7032228888@192.168.20.1:63871;transport=udp | Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) on interface any, id 0 Linux cooked capture Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128 User Datagram Protocol, Src Port: 6060, Dst Port: 4060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:7032228888@192.168.20.1:63871;transport=udp SIP/2.0 Method: INVITE Request-URI: sip:7032228888@192.168.20.1:63871;transport=udp Request-URI User Part: 7032228888 Request-URI Host Part: 192.168.20.1 Request-URI Host Port: 63871 [Resent Packet: False] Message Header Record-Route: <sip:mt@192.168.20.128:6060;lr=on;ftag=d490a30c-42a8-4fc0-9fe9-b7a21537386d;did=7d2.e0f1> Route: <sip:term@pcscf522.irisims.org;lr> Via: SIP/2.0/UDP 192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0 Via: SIP/2.0/UDP 192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1 Via: SIP/2.0/UDP 192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335 From: <sip:7033331111@asterisk.irisims.org>;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d To: <sip:7032228888@ims522.irisims.org> Contact: <sip:asterisk@192.168.20.131:5060> Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660 [Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660] CSeq: 19314 INVITE Route: <sip:ims522.irisims.org:5060;lr> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 68 User-Agent: Asterisk PBX 16.6.0 Content-Type: application/sdp Content-Length: 1121 Message Body From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 11:21 AM To: miconda(a)gmail.com<mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Martin W Woscek <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. 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 -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

1 0

Question about textops
by Mojtaba 26 Jun '20

26 Jun '20

Hello, Some confusing things while developing in Kamailio: In is_method_f function in textops module, the code checks the HDR_CSEQ_F in msg also, and if the method name appears in CSEQ_F, The result is returned TRUE. These question will arise why in this function, the CSEQ field in message will check? if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL) { LM_ERR("cannot parse cseq header\n"); return -1; /* should it be 0 ?!?! */ } if(m->s==0) return (get_cseq(msg)->method_id&m->len)?1:-1; else return (get_cseq(msg)->method_id==METHOD_OTHER && get_cseq(msg)->method.len==m->len && (strncasecmp(get_cseq(msg)->method.s, m->s, m->len)==0))?1:-1; I think it would be nice to add new function like: is_cseq_method_f in module to avoid some problem understanding. -- --Mojtaba Esfandiari.S

2 2

rtpengine inserted random line into sdp
by Andrew Chen 26 Jun '20

26 Jun '20

Hi forum, I'm starting my rtpengine project and I'm facing a strange problem with rtpengine. I am seeing this in the SDP part of the INVITE: a=rtcp:52021 a=rtcp-mux 2001:470:7:3A7:0:0:0:2 a=direction:active a=oldmediaip:54.153.25.234 As you can see there is a random insert of the local kamailio IP in there and I'm have tried the following to remove it but failed to succeed: - Remove in the offer the combination of ICE and replace-session-connection flags - Switched between rtpengine_offer and rtpengine_manage - Restarted ngcp-rtpengine-daemon process - Used sdpops function sdp_remove_line_by_preifx() Currently running Kamailio 5.1.2. I am running out of ideas. Any suggestions would be greatly appreciated. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 4

Advertise concept
by Social Boh 25 Jun '20

25 Jun '20

Hello, maybe i don't understood completely the use of Advertise parameter. We use it only if Kamailio is behind a NAT and Kamailio have to comunicate over is PublicIP with provider, other SIP proxy and PBX but never to accept Users REGISTER. Is it right? Regards -- --- I'm SoCIaL, MayBe

2 3

Server error on LIR select next S-CSCF
by BALL SUN 25 Jun '20

25 Jun '20

Hi I have a setup using Kamailio IMS, and when trying to make a SIP call between clients, S-CSCF returns " SIP/2.0 500 Server error on LIR select next S-CSCF", do you have any idea on how to fix? - RBK

2 5

kamailio - 5.3.2 - Possible memleak in Core Module
by José Seabra 25 Jun '20

25 Jun '20

Hello there. During my stress tests against our kamailio servers I detected that it was running out of shm memory. edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:298]: qm_find_free(): qm_find_free(0x7ff370920000, 4224); Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:432]: qm_malloc(): qm_malloc(0x7ff370920000, 4224) called from core: core/sip_msg_clone.c: sip_msg_shm_clone(496), module: core; Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/sip_msg_clone.c:499]: sip_msg_shm_clone(): could not allocate shared memory from shm pool edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1293]: new_t(): out of mem: edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1439]: t_newtran(): new_t failed edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) The output of kamctl stats shmem confirms that kamailio is running out of memory: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 36427", "shmem:free_size = 14123552", "shmem:max_used_size = 268435456", "shmem:real_used_size = 254311904", "shmem:total_size = 268435456", "shmem:used_size = 215787904" ], "id": 10901 } Then I was check what was the module that were consuming more memory with kamcmd mod.stats all shm and the output of this command showed me that Core module was consuming most of shm memory, as you can see bellow is the output of this command shows that Core - build_req_buf_from_sip_req is consuming most of shm. Module: core { sip_msg_shm_clone(496): 4872856 create_avp(175): 24768 msg_lump_cloner(986): 154376 xavp_new_value(106): 780896 build_req_buf_from_sip_req(2187): 183984624 counters_prefork_init(211): 36864 cfg_clone_str(130): 96 cfg_shmize(217): 848 main_loop(1303): 8 init_pt(106): 8 init_pt(105): 8 init_pt(104): 5256 register_timer(995): 232 cfg_register_ctx(47): 64 init_tcp(5021): 8192 init_tcp(5015): 32768 init_tcp(5007): 8 init_tcp(5000): 8 init_tcp(4993): 8 init_tcp(4987): 8 init_tcp(4975): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 timer_alloc(498): 96 init_dns_cache(361): 8 init_dns_cache(352): 16384 init_dns_cache(344): 16 init_dns_cache(336): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 qm_shm_lock_init(1202): 8 Total: 190229920 } Then I stopped the sipp that was running against kamailio in order to see if the memory consumed by *build_req_buf_from_sip_req* decreased but it didn't, so, it seems that there is a memleak in this version. If the information that I'm sending in this email is not enough to identify the root cause of this, please let me know if there is anything else that I can do here to help identify better what is the main reason. Than you Best Regards -- Cumprimentos José Seabra

2 2

siptrace: methods blacklist modparam
by Nuno Ferreira 25 Jun '20

25 Jun '20

Hello, We are a heavy Homer/Hepic (tks QXIP team) users and on our setup, there are some SIP methods we don’t want to get into the database or send to the capture server because there’s no value-added for call troubleshooting purposes (OPTIONS, KDMQ, HTTP (from JSONRPC)…) Typically we achieve the message filtering using siptrace in “manual” mode and control what to send from the script. This works fine for some situations, but there are a few we need to have the automatic mode on: - topos enabled: using manual mode we don’t get the “wire” version of the signaling - TCP connections: there are a few situations where the destination port is traced with 0 - I believe this is due to the fact the TCP socket not established when the message is collected There are other technics to filter like using a HEP relay proxy dropping unwanted traffic, but having a native kamailio functionality would make things much easier. With that in mind, I’ve created a PR (#2374 <https://github.com/kamailio/kamailio/pull/2374>) that allows to blacklist methods we don’t want to send to the capture server when using automatic trace mode. I believe this is a very helpful change for the community as well. To block SIP OPTIONS and KDMQ we just need to declare a new modparam: modparam("siptrace", "trace_on", 1) modparam("siptrace", "methods_blacklist_auto", 16896) Let me know what you think about it. Nuno -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working
by supreeth herle 25 Jun '20

25 Jun '20

Please try to keep the mailing list in loop so that it helps others as well facing this issue. I believe this was resolved in 5.3.3 branch not sure why you are facing this issue. Anyways, make sure that modules loaded in kamailio.cfg of PCSCF is in following order loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif loadmodule "ims_registrar_pcscf" Best Regards, Supreeth On Wed, 24 Jun 2020 at 13:36, Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi Supreeth, > > I am getting the below error > > ms_usrloc_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_usrloc_pcscf.so] > 0(455) DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: > mod_init(): Successfully bound to PCSCF Usrloc module > 0(455) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): > found export of <bind_ims_ipsec_pcscf> in module ims_ipsec_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_ipsec_pcscf.so] > 0(455) ERROR: ims_ipsec_pcscf [cmd.c:85]: bind_ipsec_pcscf(): > configuration error - trying to bind to ipsec pscscf module before being > initialized > 0(455) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while > initializing module ims_registrar_pcscf > (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_registrar_pcscf.so) > ERROR: error while initializing modules > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:305]: mod_destroy(): > Error destroying spi generator > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): > Error destroying port generator > 0(455) DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start > 0(455) DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table > 0(455) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores > 0(455) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists > > > Kindly help > > On Wed, Jun 24, 2020 at 5:03 PM supreeth herle <herlesupreeth(a)gmail.com> > wrote: > >> There is no other IPSec configuration needed apart from running Kamailio >> P-CSCF and UE sending a SIP Register with Security-Client information for >> IPSec creation. Kamailio IMS IPSec module creates IPSec tunnels using >> Netlink library. >> >> Best Regards, >> Supreeth >> >> On Wed, 24 Jun 2020 at 13:29, Pavithra Mohanraja <pavimohan3004(a)gmail.com> >> wrote: >> >>> Hi , >>> >>> Could you please tell how to configure ipsec (general configuration not >>> related to kamailio). >>> >>> Thanks, >>> Pavithra >>> >>> On Wed, Jun 24, 2020 at 2:12 PM supreeth herle <herlesupreeth(a)gmail.com> >>> wrote: >>> >>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>> >>>> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or >>>> any other HSS you are using to enter IMS user details). SCSCF is not able >>>> to find a registered user with that IMPU (i am assuming that you are >>>> getting this response while calling) >>>> >>>> Increase the debug level on S-CSCF and check out which IMPU is being >>>> referred by SCSCF >>>> >>>> Best Regards, >>>> Supreeth >>>> >>>> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja < >>>> pavimohan3004(a)gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I am facing below error in kamailio-ims 5.3.4 version. >>>>> >>>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>>> could you please help me how should i proceed with and where the >>>>> registration details will be stored in kamailio ims. >>>>> >>>>> It fails at this line in kamailio.cfg file in scscf . >>>>> >>>>> route[FINAL_TERM] { >>>>> if (lookup("location")) { >>>>> if (uri==myself) { >>>>> if (!t_newtran()) { >>>>> sl_reply_error(); >>>>> exit; >>>>> } >>>>> xlog("Destination Not Found ----\n"); >>>>> t_reply("404","Not Found - destination user >>>>> not found on this S-CSCF"); >>>>> exit; >>>>> } >>>>> >>>>> here lookup(location) means what it means. Could you please help me >>>>> >>>>> >>>>> >>>>> Thanks, >>>>> Pavithra >>>>> >>>>> On Thu, May 14, 2020 at 9:59 PM supreeth herle < >>>>> herlesupreeth(a)gmail.com> wrote: >>>>> >>>>>> Hello Pavithra, >>>>>> >>>>>> I have written a guide for how to integrate an IMS using Kamailio >>>>>> with open5gs EPC, here is the link >>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can >>>>>> find steps to setup rtpengine in that guide. >>>>>> >>>>>> Regarding requirement of rtpengine, its mandatory if you want to use >>>>>> the scripts in that example folder without modification. Also, read the >>>>>> READE of sipwise/rtpengine github repo, its very informative (in a gist >>>>>> rtpengine offer much more feature than rtpproxy) >>>>>> >>>>>> why can't we have sip client to sip client rtps flowing like how it >>>>>>> is happening without NAT. >>>>>>> >>>>>> >>>>>> For this, i would definitely suggest to read about SIP working. This >>>>>> is a perfect link https://github.com/onmyway133/blog/issues/284 >>>>>> which offers extensive material (short and very interesting to read with >>>>>> examples) related to SIP. >>>>>> >>>>>> Best Regards, >>>>>> Supreeth >>>>>> >>>>>> >>>>>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja < >>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>> >>>>>>> Hi , >>>>>>> >>>>>>> I didn't configure rtpengine ..instead of that only I used >>>>>>> rrpproxy..will it not work? >>>>>>> >>>>>>> If not please tell me how to configure rtpengine and also tell me if >>>>>>> it is really madatory .. why can't we have sip client to sip client rtps >>>>>>> flowing like how it is happening without NAT. >>>>>>> >>>>>>> That's my doubt ..please clarify .. >>>>>>> >>>>>>> Thanks. >>>>>>> >>>>>>> On Thu, May 14, 2020, 8:57 PM supreeth herle < >>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>> >>>>>>>> HI, >>>>>>>> >>>>>>>> Hold on a moment, >>>>>>>> >>>>>>>> >>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>> 10.40.10.30 - hss >>>>>>>>> >>>>>>>> >>>>>>>> So i assume you are using the examples provided in the ims folder >>>>>>>> of kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have >>>>>>>> you modified the P-CSCF configuration to work with RTPProxy? >>>>>>>> >>>>>>>> Best regards, >>>>>>>> Supreeth >>>>>>>> >>>>>>>> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja < >>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi , >>>>>>>>> I have attached working rtp pcap file. But it is getting reverted >>>>>>>>> back to me . >>>>>>>>> >>>>>>>>> As you said , i have tried restarting rtpproxy several times. but >>>>>>>>> it didnt work . >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, May 14, 2020 at 7:11 PM supreeth herle < >>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Thanks for the pcap. Sorry, to ask this but can you also send >>>>>>>>>> pcap of scenario where you see RTP packets? >>>>>>>>>> >>>>>>>>>> Also, did you restart the rtpproxy after altering the >>>>>>>>>> /etc/default/rtpproxy ? >>>>>>>>>> >>>>>>>>>> i.e. >>>>>>>>>> >>>>>>>>>> $ systemctl restart rtpproxy >>>>>>>>>> >>>>>>>>>> Best Regards, >>>>>>>>>> Supreeth >>>>>>>>>> >>>>>>>>>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja < >>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hi Supreeth, >>>>>>>>>>> I have attached pcap files . kindly check and let me know. >>>>>>>>>>> filter of rtp packets and check. >>>>>>>>>>> >>>>>>>>>>> *filter : sip || rtp* >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Thu, May 14, 2020 at 5:15 PM supreeth herle < >>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi Pavithra, >>>>>>>>>>>> >>>>>>>>>>>> Thanks for the clarification. Can you please attach a pcap of >>>>>>>>>>>> the call where you are not seeing the RTP packets? And, btw is the call >>>>>>>>>>>> getting dropped or something in that scenario? >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> Supreeth >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Thu, 14 May 2020 at 13:34, Pavithra Mohanraja < >>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi Supreeth, >>>>>>>>>>>>> >>>>>>>>>>>>> I have kamailio ims installed in 4 vms inside openstack. >>>>>>>>>>>>> >>>>>>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>>>>>> 10.40.10.30 - hss >>>>>>>>>>>>> >>>>>>>>>>>>> so all these vms are running in same subnet. >>>>>>>>>>>>> I have a floating ip associated to pcscf. >>>>>>>>>>>>> Now I have two zoiper clients outside openstack >>>>>>>>>>>>> zoiper1 - 10.0.2.15 (NAT Enabled Network adaptor) >>>>>>>>>>>>> zoiper2 - 10.0.2.15 >>>>>>>>>>>>> >>>>>>>>>>>>> when i am making a call from zoiper1 to zoiper2, call is going >>>>>>>>>>>>> properly . SIP messages are coming properly in wireshark packets. >>>>>>>>>>>>> when i filter for RTP packets in wireshark, it is not coming. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> when i have the zoiper clients inside openstack itself ,i can >>>>>>>>>>>>> able to see RTP Packets in wireshark between two sip clients without any >>>>>>>>>>>>> NAT related configurations done. >>>>>>>>>>>>> >>>>>>>>>>>>> Kindly help me. >>>>>>>>>>>>> Thanks. >>>>>>>>>>>>> >>>>>>>>>>>>> On Thu, May 14, 2020 at 4:49 PM supreeth herle < >>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Can you explain some more when you mean by "RTP alone is not >>>>>>>>>>>>>> coming."? And, how are you checking/verifying that? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:12, Pavithra Mohanraja < >>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Yes I have configured as you mentioned only . In my >>>>>>>>>>>>>>> pcscf only i have rtpproxy also configured. >>>>>>>>>>>>>>> Floating IP i have given in listen address also as >>>>>>>>>>>>>>> listen = 10.40.10.3:4060 advertise 10.45.4.22:4060 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 10.40.10.3 - private ip >>>>>>>>>>>>>>> 10.45.4.22 - floating ip >>>>>>>>>>>>>>> Still RTP alone is not coming. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Is it the only way of configuring NAT ? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Thu, May 14, 2020 at 4:37 PM supreeth herle < >>>>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hello Pavithra, >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Have you tried advertising the floating IP of the VM >>>>>>>>>>>>>>>> running RTPProxy as follows? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Edit /etc/default/rtpproxy file as follows: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> # Additional options that are passed to the daemon. >>>>>>>>>>>>>>>> EXTRA_OPTS="-l 172.24.15.30 -d DBUG:LOG_LOCAL0" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> here, "-l <PUBLIC_IP>" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hope it helps. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:01, Pavithra Mohanraja < >>>>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Could anybody please give me the approach how to configure >>>>>>>>>>>>>>>>> NAT in kamailio IMS. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I have installed kamailio IMS in openstack environment >>>>>>>>>>>>>>>>> which has private ips for all four vms (each component in one vm). >>>>>>>>>>>>>>>>> pcscf has one floating IP attached. I have enabled "WITH_NAT" and >>>>>>>>>>>>>>>>> configured rtpproxy . >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> when i make a call from outside of openstack using 2 >>>>>>>>>>>>>>>>> zoiper clients with outbound proxy keeping floating ip of pcscf, I am >>>>>>>>>>>>>>>>> getting 200 ok with all SIP packets correctly flowing but there is no RTP >>>>>>>>>>>>>>>>> Packet. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> When i try the same within openstack environment without >>>>>>>>>>>>>>>>> using public ip, I am getting RTP Packets. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly tell me the procedure of NAT Enablement or do i >>>>>>>>>>>>>>>>> need to do anything other than this. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly help because i am stuck with this for long time. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>> 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 >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>> 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 >>>>>>>>>>>>>> >>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> 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 >>>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> 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 >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> 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 >>>>>> >>>>>

2 5

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. 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 -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

1 1

Kamailio IMS using returned SRV records for transport slelection
by BALL SUN 24 Jun '20

24 Jun '20

Hi All I am new to Kamailio IMS, and I just setup a testing environment where having P-CSCF, I-CSCF and S-CSCF. I would like to use make below works 1. SIP client <-> P-CSCF (either UDP or TCP, based on the transport tag) 2. P-CSCF <-> I-CSCF (TCP by default) 3. I-CSCF <-> S-CSCF (TCP by default) For point 2, I observed that P-CSCF actually will send DNS SRV query to DNS server for resolving _sip._udp. and _sip._tcp. I believed we can just disable the "_sip._udp." so that P-CSCF will always use TCP to communication wih I-CSCF. However, for point 3, I didnt notice I-CSCF send DNS SRV query, My question is how we can force I-CSCF to do that or is there any configuration that we should enable such that I-CSCF always use TCP to S-CSCF? - RBK

2 2

SECFILTER reload from Siremis
by Social Boh 24 Jun '20

24 Jun '20

Hello list, is posible to reload SECFILTER data directly from Siremis? or the only way is: kamcmd secfilter.reload Regards -- --- I'm SoCIaL, MayBe

3 3

dialog's track_cseq_updates seems to break uac_auth() in this configuration
by George Diamantopoulos 24 Jun '20

24 Jun '20

Hello again, This might turn out to be a bug, but I'm posting here first because my config is fairly complicated and it might not be the sanest it could be. The problem manifests as follows: with the sole change being the enabling of track_cseq_updates for the dialog module, kamailio fails to process the 100 Trying it receives following authentication initiated by the uac_auth() function. Without the track_cseq_updates option enabled, the call proceeds successfully, albeit with the CSeq for the auth-carrying INVITE having the same value as the original. Here's a simplified version of the configuration routes involved in this: request_route { ... route(DISPATCH_PROVIDER); ... } route[DISPATCH_PROVIDER] { if ( !ds_select_dst("10", "8") ) { t_send_reply("503", "Downstream carrier unavailable"); exit; } t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_branch("PROVIDER_BRANCH"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } failure_route[DISPATCH_PROVIDER_FAILURE] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(arealm) = "authrealm.com"; $avp(auser) = "authusername"; $avp(apass) = "verys3kr1t"; uac_auth(); t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll explain why this is here later in this e-mail t_relay(); exit; } } branch_route[PROVIDER_BRANCH] { uac_replace_from("$dlg_var(my_new_from)"); uac_replace_to("$dlg_var(my_new_to)"); $rU = "my new RURI user"; } event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { return; # next DST - only for 5xx or local timeout } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout() && !t_branch_replied()) ) { if ( ds_next_dst() ) { t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } else { xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER: Failed to route request to PROVIDER! Giving Up. Negative response will be sent upstream.\n"); } } } One will rightfully wonder why use both branch-failure routes and failure_route to handle things. Well, it's not obvious from this because the relevant parts are removed for brevity, but I'm generally using branch failure event routes to try other destinations in the same destination set, and failure routes to additionally try other downstream "providers" in case all entries is the current destination set have failed. I don't think these actions/routes are relevant because the problem manifests without any of these failover mechanisms (of switching over to the next provider) engaging. Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);" in the failure_route, which might seem a little odd, this was added because without it, uac_auth() will send the auth-carrying INVITE always to the first destination in the destination set even if ds_next_dst has been called from the branch-failure event route. Unfortunately I haven't been able to determine if this actually helped, because I have only received a 5xx error once from the downstream peer before the "fix" and haven't been able to reproduce this since then. But that's another issue unrelated to the one I'm asking about here. So what happens with track_cseq_updates enabled is, after uac_auth() successfully sends the authenticated INVITE and the peer starts sending provisional responses, kamailio doesn't seem to acknowledge them. Instead, it will retransmit the auth INVITE as if there was a firewall preventing the 1xx responses from being admitted. Then, the timeout will engage and the configuration script will behave as if t_branch_timeout returns true (it will do ds_next_dst). So I was wondering if this is to be expected with this configuration, or if this should be reported as a bug. Thanks! Best regards, George

2 2

Kamailio-5.3.3. NAT and RTP not working
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi , Could anybody please give me the approach how to configure NAT in kamailio IMS. I have installed kamailio IMS in openstack environment which has private ips for all four vms (each component in one vm). pcscf has one floating IP attached. I have enabled "WITH_NAT" and configured rtpproxy . when i make a call from outside of openstack using 2 zoiper clients with outbound proxy keeping floating ip of pcscf, I am getting 200 ok with all SIP packets correctly flowing but there is no RTP Packet. When i try the same within openstack environment without using public ip, I am getting RTP Packets. Kindly tell me the procedure of NAT Enablement or do i need to do anything other than this. Kindly help because i am stuck with this for long time. Thanks.

4 26

multiple domains with microsoft team
by Leanderson Correia 24 Jun '20

24 Jun '20

I'm testing here to use multiple domains with microsoft teams and I'm having trouble making and receiving calls. It seems that when connecting a call, kamailio keeps the tls connection active and reuses it, making the domain parameters used the wrong ones for new call. Can someone help me, or give a tip? -- Att, Lean

3 2

Presence and call queueing - Kamailio and Asterisk
by Pover A.V. 24 Jun '20

24 Jun '20

Summary: Using Kamailio to load balance between Asterisk servers. Trouble getting call queuing working because can't get device state hints to work reliably. - Hello I'd be grateful for any advice that can be offered for this problem to do with presence and call queueing. I have 3 Kamailio (v5.3.4) servers that load-balance incoming traffic from various sources: a legacy PBX, soft phone clients, and a SIP trunking service. Sitting behind the 3 Kamailio servers are 2 Asterisk (v16 LTS) servers. Kamailio acts as a registrar and user location service. Asterisk is routing calls and (should be) providing device state information for each extension. An external script builds a list of PJSIP endpoints based on the content of the Kamailio subscriber table, and deploys it to the Asterisk servers whenever a change is detected. The AOR for each record is configured to point to Kamailio. No authentication details are attached to these endpoints, as Kamailio is doing authentication before dispatching a request to Asterisk. Whenever Kamailio receives an INVITE, it passes this to an Asterisk server using the dispatcher module (round-robin algorithm '4'); asterisk then processes the request. At the moment, everything is working except for presence/device states and call queueing. I've tried to set up 'presence', 'pua', 'dialog' and '(presence/pua)_dialoginfo' modules on Kamailio with limited success. The presence state when a user toggles 'do not disturb' on their soft phone is passed to other users who are subscribed (watching), however when a user makes a call, no presence updates are seen by subscribed users (the extensions show as 'Idle' or 'Available' instead of 'In Use' or 'Busy'). Asterisk is able to report that an extension is 'Ringing' or 'InUse' when a call is made to that extension, because an INVITE to an internal extension matches a pattern in the dialplan and Asterisk routes it to a PJSIP endpoint with the same name. However, when somebody makes a call, Asterisk is unable to track the state of the extension making the call because the call is going via Kamailio so Asterisk is not seeing the caller as an internal extension. We need a way to make Asterisk think the call is coming from an internal device if the caller ID matches a PJSIP endpoint. Both Asterisk servers are configured to share device state updates with each other, and this is working. Without being able to see device state for an internal calling party, we will be unable to reliably set up call queues in Asterisk. Using the 'mohqueue' module on Kamailio is not something we want to do because of its reliance on a database, which is creating a single point of failure - if the database server is down for any reason, Kamailio refuses to start. Ideally, we would achieve some form of distributed call queueing, however the simplest option seems to be setting up a queue on one Asterisk server and having an active-passive high availability configuration, should a server go down. I'm looking for advice on how best to keep track of what is happening in a call, and for Asterisk to know the states of each extension and make these available in a dialplan hint. Or any other suggestions that might help to achieve call queueing in this setup. Thank you

2 1

Dialog Profiles
by Ilie Soltanici 24 Jun '20

24 Jun '20

Hello, Is there any way in Kamailio to get the profile values the current dialog is part of? For example, I have one dialog with the following profiles: profiles: { { test: abcd } test: 1234 } I need in the reply route to get the values for the test dialog profile. (abcd, 1234). I know I can use get_profile_size, but that will give the number of profiles of the current dialog, but I need the values to loop through. Thanks.

2 2

Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin

2 1

Kamailio ims video calls
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi, Could anyone please give me some idea on how video call works in IMS. Does it need rtp proxy to work? Or is it the same way how voice call works ? It would be good , if anybody share some links on video calls as well voice conferencing. Thanks, pavithra

2 1

kamailio 5.2.2 enabling hashing_type
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello, I'm revisiting some initial issues I had with P-CSCF. I could not set the hashing type in my PCSCF (kamalio 5.2.2) so I simply commented it out and moved on. But now, I may need it as the P-CSCF is not handling INVITES coming from I-CSCF that originated from external SIP server, where the endpoint (UE) is registered to the IMS but the P-CSCF sends the INVITE back to the S rather that the registered UE and fails. kamailio.cfg: modparam("ims_usrloc_pcscf", "hashing_type", 1) So with that, when I set the hashing_type to 1,1 or 2 I get the following starting the pcscf: Starting the P-CSCF: Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Not starting Kamailio IMS P-CSCF: invalid configuration file! Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * 0(5766) ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: 0(5766) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 364, column 47: Can't set module parameter Am I missing some module that I did not include at compile time? According to the docs, it has no additional dependencies other than the ims_usrloc_pcscf.so https://kamailio.org/docs/modules/5.0.x/modules/ims_usrloc_pcscf.html#idp35… Any help is appreciated. Thanks, _Martin

2 1

kamailio5.2.2 calling t_relay(host, port)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, Im trying to call t_relay() with host and port as documented in: https://www.kamailio.org/docs/modules/5.2.x/modules/tm.html However, I get a subst_str(): no match on P-CSCF start and therefore fails. Is t_relay(host, port) supported as documented? Am I missing something? Thanks, _Martin

2 1

Expired transaction
by Marat Gareev 24 Jun '20

24 Jun '20

Hello! I'm trying to manually reproduce expired transactions which couldn't be cleared. I had one idea, but it was unsuccessful. I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared. Do you have any ideas for reproduction? P.S. I'm sorry If you received my incoherent mails. These were attempts to reply to the thread, seems to be my mail client or sr-users mailing list doesn't support In-Reply-To header.

2 1

ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found
by Denys Pozniak 23 Jun '20

23 Jun '20

Hello! Could someone point me to the possible source of the problem? Jun 16 09:51:06 proxy-edge /usr/sbin/kamailio[9645]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9640]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9600]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9660]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Modules settings: *loadmodule "pua.so"* *loadmodule "pua_dialoginfo"* *loadmodule "dialog.so" * *modparam("pua", "default_expires", 1200)* *modparam("pua", "outbound_proxy", LOCAL_IP_ADDR)* *modparam("pua_dialoginfo", "include_callid", 1)* *modparam("pua_dialoginfo", "include_localremote", 1)* *modparam("pua_dialoginfo", "include_tags", 1)* *modparam("pua_dialoginfo", "send_publish_flag", 8)* *modparam("dialog", "dlg_flag", 4)modparam("dialog", "db_mode", 0)modparam("dialog", "enable_stats", 1)modparam("dialog", "dlg_match_mode", 1)modparam("dialog", "default_timeout", 7200) modparam("dialog", "hash_size", 8192)* *request_route{* *...* *setflag(8);* *...* *}* Generated SIP PUBLISH: *PUBLISH sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com> SIP/2.0* *Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bK3e55.afdce977000000000000000000000000.0* *To: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>* *From: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>;tag=e98d9fc94bada9746d67946ab287a76f-ba77* *CSeq: 10 PUBLISH* *Call-ID: 72ef85f374c8e703-9659(a)10.10.10.10 <72ef85f374c8e703-9659(a)10.10.10.10>* *Content-Length: 578* *User-Agent: proxy* *Max-Forwards: 70* *Event: dialog* *Expires: 7201* *Content-Type: application/dialog-info+xml* *<?xml version="1.0"?>* *<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>">* * <dialog id="162092997-5060-464(a)BJC.BGI.CA.BBI" call-id="162092997-5060-464(a)BJC.BGI.CA.BBI" direction="initiator">* * <state>Trying</state>* * <remote>* * <identity>sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com></identity>* * <target uri="sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com>"/>* * </remote>* * <local>* * <identity>sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com></identity>* * <target uri="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>"/>* * </local>* * </dialog>* *</dialog-info>* -- BR, Denys Pozniak

2 5

icscf start error
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I have installed kamailio - 5.3.4 version where i am facing below issues in icscf while starting kamailio. db_mysql is loaded but .. what could be the issue? icscf : Aliases: *: ims.sip.example.com:* 0(789) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(791) INFO: pike [pike.c:97]: pike_init(): PIKE - initializing 0(791) INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size *0(791) ERROR: <core> [db.c:204]: db_bind_mod(): Module db_con1=>mysql not found. Missing loadmodule? 0(791) ERROR: ims_icscf [db.c:83]: ims_icscf_db_bind(): ims_icscf_db_bind: cannot bind to database module! Did you forget to load a database module ? 0(791) ERROR: ims_icscf [ims_icscf_mod.c:270]: mod_init(): Failed to initialise ims_icscf module 0(791) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_icscf (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_icscf.so)* ERROR: error while initializing modules 0(791) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... 0(791) INFO: cdp [diameter_peer.c:367]: diameter_peer_destroy(): destroy_diameter_peer(): Terminating all children... 0(791) INFO: cdp [diameter_peer.c:386]: diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated. Cleaning up. 0(791) INFO: cdp [worker.c:139]: worker_destroy(): Unlocking workers waiting on empty queue... 0(791) INFO: cdp [worker.c:142]: worker_destroy(): Unlocking workers waiting on full queue... 0(791) CRITICAL: cdp [diameter_peer.c:426]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test 0(791) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped 0(791) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

2 3

Kamailio v5.3.5 Released
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, Kamailio SIP Server v5.3.5 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.5. For more details about version 5.3.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/06/kamailio-v5-3-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

reason=404 (Not Found - destination user not found on this S-CSCF)
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I am using kamailio ims-5.3.4 where i am facing this kind of problem during route(term). call registration is happening successfully. In call Establishment , during route(term) , I am facing destination user not found . what could be the problem?. can anyone please help me. Thanks, Pavithra

2 2

Releasing Kamailio v5.3.5
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, I am considering to release next week (likely on Monday, June 22) the next minor version out of branch 5.3, respectively 5.3.5. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Re: [SR-Users] 403 domain not served error in scscf
by David Villasmil 20 Jun '20

20 Jun '20

you shouldn't write to me directly. You can write to the list and address me. Anyway, kamailio thinks the message received is NOT directed to it. You should add an "alias" with the IP you're sending to (or the FQDN) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Sat, Jun 20, 2020 at 9:01 PM Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi David, > Could you please help me in this issue > > ---------- Forwarded message --------- > From: Pavithra Mohanraja <pavimohan3004(a)gmail.com> > Date: Fri, Jun 19, 2020, 1:36 PM > Subject: 403 domain not served error in scscf > To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > > > Hi, > I am using kamailio IMS - 5.3.4 > when I am making a call between two clients using zoiper , Call > establishment is done successfully . > > But during BYE packet sent from caller to callee , it is reaching till > S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. > > > I have sent lot of mails regarding this error and no replies .. Kindly > help .. > > Error in scscf : > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed > from core: core/rvalue.c: rval_new_empty(224) > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && > [*10.45.5.22] == [10.45.4.181]* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 > (advertise 0) matches port 40629* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/forward.c:412]: *check_self(): host != me* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: > sip:alice at sip.example.com > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* > Domain not served 403* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( > sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, > y-B071ZDjFS-z4NLxREyVw..) > > > Kindly help .. > 10.45.5.22 - sip client alice(a)sip.example.com > 10.45.4.181 - scscf > > > please help.. >

2 1

How do I test the Kamailio service ？
by zhaosl＠shinetechchina.com 20 Jun '20

20 Jun '20

Hi : Now I set up a Kamailio Service on the server, it works now. But I didn`t know how to test the server by trying the voice tool.Can you give me some advices , tools or methods ? Thanks. zhaosl(a)shinetechchina.com

3 2

Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

Hi can you share your kamaili config. I want to add some extra columns to cdr table like what you did. Thanks. Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

2 1

Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

1 0

403 domain not served error in scscf
by Pavithra Mohanraja 19 Jun '20

19 Jun '20

Hi, I am using kamailio IMS - 5.3.4 when I am making a call between two clients using zoiper , Call establishment is done successfully . But during BYE packet sent from caller to callee , it is reaching till S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. I have sent lot of mails regarding this error and no replies .. Kindly help .. Error in scscf : Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && [*10.45.5.22] == [10.45.4.181]* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 (advertise 0) matches port 40629* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/forward.c:412]: *check_self(): host != me* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: sip:alice at sip.example.com Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* Domain not served 403* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, y-B071ZDjFS-z4NLxREyVw..) Kindly help .. 10.45.5.22 - sip client alice(a)sip.example.com 10.45.4.181 - scscf please help..

1 0

Development for v5.4.x is frozen
by Daniel-Constantin Mierla 19 Jun '20

19 Jun '20

Hello, the formal notification that the development for the next major version 5.4.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.4 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, pull requests can still be done, but they will be merged after branching 5.4. Can still be done commits with documentation improvements, enhancements to related tool (e.g., kamctl, kamcmd), exporting missing KEMI functions and completing the functionality of the new modules added in 5.4. Once the branch 5.4 is created, new features can be pushed again to master branch as usual. From that moment, the v5.3.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.4 is going to be built at: * https://www.kamailio.org/wiki/features/new-in-5.4.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.3.x-to-5.4.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.3.x to 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

MD5 and SHA-256 instead of MD5 or SHA-256...
by Aymeric Moizard 18 Jun '20

18 Jun '20

Hi All, I'd like to improve my setup by switching to SHA-256. However, as a first step, I would like to offer both MD5 and SHA-256 in 2 different WWW-Authenticate header. If I'm correct, this is not doable with the latest auth module? Is this a planned feature? As an alternative, I would like to decide the algorithm in the script instead of a module parameter. It looks to me this is also not doable? Again, is this a planned feature? Thanks to all, Regards Aymeric -- Antisip - http://www.antisip.com

5 10

Freezing development for Kamailio v5.4.0
by Daniel-Constantin Mierla 18 Jun '20

18 Jun '20

Hello, based on the discussions during the last online devel meeting, the plan is to release the next major version (Kamailio v5.4.0) around the mid of this summer. Therefore I propose to freeze the development on Thursday, June 18, 2020. Then we will have about 4-6 weeks of testing before the release is out. Master will be unfrozen when the branch 5.4 is created (usually that was done after 3-4 weeks in the testing phase). If you are planning to introduce new features in the 5.4 release series, be sure you push them to master branch or create a pull request till the freezing date. If anyone else wants a different time line for 5.4.0 release, let us know and we can discuss to match the best for everyone. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 4

removing xavp from the stack.
by David Villasmil 18 Jun '20

18 Jun '20

Hello guys, if i do something like: $(xavp(my_var)) = "1"; $(xavp(my_var)) = "2"; $(xavp(my_var)) = "3"; it is store like: $(xavp(my_var)) = "1"; -> index 2 $(xavp(my_var)) = "2"; -> index 1 $(xavp(my_var)) = "3"; -> index 0 and later i need to use and remove from the stack, like this: use and remove: $(xavp(my_var)) --> would remove index 0 Then the stack would remain like: $(xavp(my_var)) = "1"; -> index 1 $(xavp(my_var)) = "2"; -> index 0 Simply popping the index 0 and reindex the remaining... is this possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

dispatcher seems to use the default client certificate
by Mack Hendricks 18 Jun '20

18 Jun '20

Hey All, I'm attempting to use dispatcher to send probe messages using TLS for two different domains. I'm providing the socket attribute, which maps to a certificate in /etc/kamailio/tls.cfg. But, it seems to always select the default client cert, which is not the certificate I want to use. My attrs column in dispatcher looks like this: socket=tls:142.93.159.231:5061;ping_from=sip:mack.dopensource.com socket=tls:142.93.159.231:5062;ping_from=sip:levin.dopensource.com Is there some way to force dispatcher to do TLS cert matching based on the host:ip? Thanks -Mack

5 5

Remove a Line from SDP
by egemen ulus 18 Jun '20

18 Jun '20

Hi all, I am trying to remove a line from the SDP body of INVITE on kamailio 4.4 via SDPOPS module but it does not work even though there is no any error message. This is how I configure kamailio.cfg, it matches with if condition and executes "sdp_remove_line_by_prefix" function, but when I check sended INVITE, I still see the line that beginning with "a=rtpmap". I thought, in the RELAY function, the sdp body might not prepared yet? Is there any suggestion to remove a line from the SDP body? route[RELAY] { ... ... if(is_method("INVITE")) { if( $ct =~ "0553847<tel:+10553847>aaaa"){ xlog("L_WARN", "$ct check_this \n"); sdp_remove_line_by_prefix("a=rtpmap"); } } } /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=614 a=16 n=if /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=611 a=26 n=xlog /path/kamailio[10332]: WARNING: <script>: <sip:0553847<tel:+10553847>aaaa@x.x.x.x:5060> check_this /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=612 a=25 n=sdp_remove_line_by_prefix /path/kamailio[10332]: exec: *** cfgtrace:request_route=[RELAY] c=[/path/kamailio.cfg] l=622 a=16 n=if Regards Egemen U. Get Outlook for Android<https://aka.ms/ghei36>

2 2

null branch(count) variable
by Ilie Soltanici 18 Jun '20

18 Jun '20

Hello, As per documentation after lookup() function I have to get the number of branches from the main branch by using $branch(count) variable, but for some reason, I'm getting always only <null> value, can someone help me to understand what I'm doing wrong? lookup(); xlog("Branch Count: $branch(count)\n"); Kamailio Version: 5.2.7 Thanks.

3 4

403 domain not served error in IMS kamailio
by Pavithra Mohanraja 18 Jun '20

18 Jun '20

Hi, Could any one please help me in this issue ASAP. This is logs from SCSCF. Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 10==10 && [172.17.0.8] == [172.17.0.5] Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/socket_info.c:637]: grep_sock_info(): checking if port 4080 (advertise 0) matches port 5060 Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:172.17.0.8:5060) vs. (0: sip.example.com:0) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:172.17.0.8:5060) vs. (0: sip.example.com:5060) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:172.17.0.8:5060) vs. (0: scscf.sip.example.com:4080) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [*core/forward.c:423]: check_self(): host (0:172.17.0.8:5060 <http://172.17.0.8:5060>) != me* Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: ERROR: <script>: sip:bob at sip.example.com *Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: ERROR: <script>: Domain not served 403* Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: ERROR: <script>: BYE ( sip:alice@sip.example.com (172.17.0.3:4060) to sip:bob@sip.example.com, -KT6eSWl7Dm77fe5hbNbeiHR2kwBjni0) Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7fd1f6db5010, 30) called from core: core/ut.c: as_asciiz(161) scscf.cfg # SIP / UDP listen=udp:172.17.0.5:4080 # SIP / TCP listen=tcp:172.17.0.5:4080 # SIP / TCP/TLS listen=tls:172.17.0.5:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 alias=sip.example.com:5060 alias=sip.example.com # ENUM-Server to query:

1 0

docker images for kamailio IMS
by Pavithra Mohanraja 18 Jun '20

18 Jun '20

Hi , Could anyone kindly tell me if there is docker images for kamailio IMS. I can able to see docker images for only Kamailio as a SIP server. Please help. Thanks, Pavithra

4 10

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users