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47 participants
37554 discussions

Re-using an existing TLS connection
by Matthias Urlichs 25 Feb '22

25 Feb '22

Hello, My problem: our provider opens a TLS connection to us. They want us to use this channel for outgoing calls, instead of opening a new SIP connection to their server. Is there a way to teach Kamailio to do that? -- -- Matthias Urlichs

3 3

Info: SIP resources collection
by Daniel-Constantin Mierla 24 Feb '22

24 Feb '22

Hello, over the 20 years I used various SIP applications, libraries and tools but after a while I forgot many of them, therefore I tried to collect many in a list hosted on github at: * https://github.com/miconda/sip-resources Hoping it proves useful for some of you! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

4 6

kamailio : how to xlog the value of "myself" ?
by Youssef Boujraf 24 Feb '22

24 Feb '22

Dear I 'd like to xlog the value of the "myself" ? if(uri==myself) { log("the request is for local processing\n"); };Best Regards, -- Youssef BOUJRAF

3 3

dispatcher.reload
by David Villasmil 24 Feb '22

24 Feb '22

Hello guys, We're dispatching with hash over user, if i execute dispatcher.reload <http://www.kamailio.net/docs/modules/5.0.x/modules/dispatcher.html#dispatch…> does the dispatcher list gets calculated? i.e. user A is always dispatched to box3, if i reload, is the hash recalculated and will probably be dispatched differently? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 4

Dispatcher Load Balance with SIP Trunk
by warawich 24 Feb '22

24 Feb '22

Hello, I would like to do a load balance between Asterisk SIP trunks. You can see a diagram from this link: https://drive.google.com/file/d/1Qy66L5rQCfxysYQpSd2-ek_8-by0T8PR/view?usp=… SIP Packets capture log: https://drive.google.com/file/d/1CHGUOwoRDAC93MMBtyfa8gBISKIVC-ng/view?usp=… Details 1. Asterisk1 makes SIP trunk connection with Kamailio. 2. Kamailio makes SIP trunk connection with Asterisk2 and Asterisk3 3. Caller register SIP phone with Asterisk1 Caller extension = 8002 4. Asterisk2 makes a blind transfer to Kamailio. (Call to 8009 then transfer to ARI APP) exten = 8009,1,Transfer(SIP/3802@<kamailio public IP>) 5. ARI app extension is 3802 My problem is that Kamailio LB only works when I try to connect with Asterisk1 and Asterisk2 or Asterisk1 and Asterisk3. If I have two Asterisks in dispatcher.list, it doesn't work and it appears *SIP/2.0 401 Unauthorized* in sip packets capture log. Kamailio version: 5.5.2 Kamailio.cfg modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 3) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } dispatcher.list # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attrs(str,opt) 1 sip:10.148.0.31:5123 1 sip:10.148.0.44:5123 Thanks, Warawich

2 2

SQLOPS sql_query_async() not functioning in an async manner?
by Ben Kaufman 23 Feb '22

23 Feb '22

I'm not sure if this is by design or not. I have things I want to log to a SQL DB, but it's not nearly as important as call handling, so the sql_query_async() function looked ideal, but in testing, if the DB is not responsive, the call processing gets blocked. In the example below, I'm using just a short hostname which isn't resolvable, but if I have an IP address there that isn't really assigned (for example if the DB server is down), the same blocking behavior is exhibited, but the error is slightly different (can't connect instead of unknown host). Here's my config. This specific simple config was tested on Kamailio 5.5, but I observe similar behavior (in a more complicated config) in the dev branch as well. - - - - #!KAMAILIO async_workers=2 loadmodule "db_mysql" loadmodule "sqlops" loadmodule "sl" loadmodule "xlog" loadmodule "pv" force_rport=yes ## Host sql does not resolve. modparam("sqlops","sqlcon","cb=>mysql://kamailio:kamailiorw@sql/kamailio") ## But start anyways. modparam("sqlops", "connect_mode", 1) request_route { sl_send_reply("100", "Trying"); ## Prevent re-transmissions from the UAC xlog("L_NOTICE", "Time before sql_query_async(): [$TV(Sn)]\n"); sql_query_async("cb", "INSERT INTO table1 (col) VALUES ('val')"); xlog("L_NOTICE", "Time before sl_send_reply(): [$TV(Sn)]\n"); sl_send_reply("404", "Not found"); } Here's the log if I send a simple INVITE with SIPp. Note the timestamps 9(16) NOTICE: <script>: Time before sql_query_async(): [1645574694.546165] 9(16) ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'sql' (-3) 9(16) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 9(16) ERROR: sqlops [sql_api.c:190]: sql_reconnect(): failed to connect to the database [cb] 9(16) ERROR: sqlops [sqlops.c:232]: sql_check_connection(): failed to connect to database 9(16) ERROR: sqlops [sqlops.c:264]: sql_query_async(): invalid connection to database 9(16) NOTICE: <script>: Time before sl_send_reply(): [1645574699.552440] Ben Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

4 4

kamailio :?==?utf-8?q? ROUTE[AUTH] - IPAUTH - use case
by Youssef Boujraf 23 Feb '22

23 Feb '22

Dear, I enabled two softphone with ADDRESS. - codec1 : ip 192.168.30.241/32 5060 - codec2 : ip 192.168.30.242/32 5060 I enable one softphone with SUBSCRIBER - codec3 : is registered (login/passwd) codec4 is not part of the domain, is not subriber and address. Kamailio proxy - kamailio : ip 192.168.30.240/32 5060 In the kamailio.cfg ROUTE[AUTH]# IP authorization and user authenticationroute[AUTH] {#!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return;} Regarding the condition #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif USE CASE : If codec4 (not registered or same domain) tries to INVITE codec3 (in subscriber) the codec, an INVITE MESSAGE is sent and the codec3 could ACK. - But I don't want to allow the INVITE, I 'd like to not allow the codec4 to reach codec3. - But if codec1 or codec2 tries to reach codec3, kamailio will allow the INVITE. Best Regards, -- Youssef BOUJRAF

1 0

Issue with Authenticating outbound Call
by Victor ku 22 Feb '22

22 Feb '22

Hi, I'm new to sip and kamailio, I have recently setup a docker container based on kamailio/kamailio:5.5.0-stretch to check things out. While I was trying to authenticate an outbound call to a 3rd party sip provider from my kamailio container, I noticed there are times that the 2nd invite with the authorization header that's supposed to be sent from the failure route doesn't get sent out. There's no error recorded in the log nor does wireshark picking up any reply to sip phone or message to 3rd party sip provider. Anyone know what might have caused this? Some other things to note: Kamailio does sent out the INVITE with authorization header and authenticated successfully when my sip phone sent the sip request with FROM ip set to the 3rd party sip server's ip. When I change the sip phone to send INVITE with FROM Domain using kamailio container's ip, I'm able to see header is being built from the trace but no INVITE with authorization header is being sent. Changing the FROM uri using $fu or uac_replace_from() does help either. ---------------------------------------------------------------------------------------------------------------- > > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} uac [auth_hdr.c:408]: > build_authorization_hdr(): hdr is <Authorization: Digest > username="xxxxxxxx", realm="xxx.xxx.xxx.xxx", nonce="07f4a5779568", > uri="sip:+0123456789@xxx.xxx.xxx.xxx", qop=auth, nc=00000001, > cnonce="3782993246", response="d5ebd0ef00d44db2292c7b8508fa23bc", > algorithm=MD5 > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1026 a=16 > n=if > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1021 a=24 > n=t_relay > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} <core> [core/msg_translator.c:1797]: > check_boundaries(): no multi-part body > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1030 a=2 > n=exit > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483650 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1564]: > t_should_relay_response(): rps discarded - uas status: 100 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1924]: relay_reply(): > reply status=2 branch=0, save=0, relay=-1 icode=0 msg status=401 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:596]: > receive_msg(): reply-route executed in: 1753 usec > kamailio-kamailio-1 | 14(22) DEBUG: dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483652 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:606]: > receive_msg(): cleaning up ---------------------------------------------- package capture ---------------------------- 192.168.0.165 xxx.xxx.xxx.xxx SIP/SDP 1383 Request: INVITE > sip:+0123456789@ xxx.xxx.xxx.xxx | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 585 Status: 100 Trying | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 820 Status: 401 Unauthorized | > 192.168.0.165 xxx.xxx.xxx.xxx SIP 396 Request: ACK sip:+0123456789 > @ xxx.xxx.xxx.xxx | ------------------------------------------nothing more till sip phone cancel ----------------- # ----------------- below is failure route config --------------------- > failure_route[MANAGE_FAILURE] { > xlog("~~~~~ MANAGE_FAILURE ($rm) ($rs-$rr) (from $fu) (to $tu) (req :$ru) > (dest $du)\r\n"); if (t_is_canceled()) { exit; } xlog("Checking status code\r\n"); #!ifdef WITH_UAC if(t_check_status("401|407")) { if (isflagset(7)) { t_reply("503","Authentication failed"); exit; } $avp(auser) = "xxxxxxxxxx"; $avp(apass) = "xxxxxxxxxx; if(uac_auth()) { setflag(7); xlog("after uac_auth $rm (from $fu)(to $tu)(req $ru)(dest $du)\r\n"); if(!t_relay()) { xlog("Failed to relay ($rm) ($rs-$rr) (from $fu)(to $tu)(req $ru)\r\n"); t_reply("500","Error"); } } else { xlog("uac_auth() not available\r\n"); t_reply("503","Authentication invalid"); } exit; } #!endif } > # ----------- end failure route config --------------------

2 2

reinvite with 2 video lines in SDP
by Filippo Graziola 22 Feb '22

22 Feb '22

Hi all, I'm facing a strange behaviour from a client (iOS application based on pjsip library) which is doing as follow when receiving a call (on phone locked): - accept the call rejecting video (m=video 0...) in SDP - reinvite to negotiate again video media Specifically the reinvite passing through kamailio (+rtpengine) has this SDP body after rtpengine offer (the client use TLS so I can't see the original message): v=0 o=- 3854443139 3854443142 IN IP4 PUBLIC_RTP_ENGINE_IP s=pjmedia b=AS:386 t=0 0 a=X-nat:0 m=audio 46734 RTP/AVP 8 101 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:96000 a=ssrc:1456418416 cname:441b90ac6db72f22 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:46735 m=video 0 RTP/AVP 96 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 m=video 46776 RTP/AVP 98 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=42e01e; packetization-mode=1 a=sendrecv a=rtcp:46777 from this I can understand that the client is doing something strange, because there is still the rejection line (I know that this question should go to the pjsip guys) but I would also like to know if it is possible to "clean" this sdp by kamailio. A few clients receiving this sdp are answering with only audio. Reading the docs, sdpops/textops modules allow modifying lines by prefix or media, but till now I have no luck to exclude only the line "m=video 0 RTP/AVP 96" and his attributes without touching the correct video request. Could someone please give me some hints on it?

2 2

Enable switch redirect to other SIP server source
by bayu p 21 Feb '22

21 Feb '22

Dear Kamailio Expert, Could you help me switching redirect sip server to another sip server ? As described below architecure. I have some uac_redirect module to implement , but kamailio cannot perform running states. Pleas help to reditect sip server to another sip server. Thanks & Best regards,

2 1

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