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13 participants
37464 discussions

Does pure outbound RFC5626 work without edge proxies?
by Richard Chan 17 Jul '22

17 Jul '22

Hi sr-users, Does pure RFC5626 outbound work without an edge proxy->internal proxy design? By "pure" I mean no nathelper to rewrite addresses. I tried setting up the following: Configuration: use standard kamailio configuration without WITH_NAT (Ignore RTP-rtpengine/rtpproxy for the moment as I am just testing the control plane) 1. UA1 UA2 each behind NAT, with bogus Contact: headers 2. kamailio loaded with rr/tm (no outbound/path because I am using single proxy/registrar — without edge proxies) 3. no nathelper: no Contact rewriting, set_contact_alias etc 4. Both UA1 and UA2 establish persistent TLS to kamailio 5. INVITE/200 OK/ACK does work. 6. Contact headers show the NAT bogus addresses (as expected). UA1/UA2 are configured to be "dumb"(i.e. don't self-rewrite Contact:/SDP based on rport/received). Problem: 7. In-dialog requests don't work as kamailio tries to route to the bogus addresses per the Contact header. I.e. kamailio doesn't seem to resolve the UAs(bogus Contact addresses) to the persistent TLS connections. Now in the Route header I don't see any flow-token so probably kamailio is doing "normal" routing instead of RFC5626 routing (i.e. because it has no flow-token to match with an existing flow). This leads me to ask: for pure RFC5626 to work (no nathelper stuff at all) is it a must that there is a "edge-proxy--[1]-->internal-proxy" layout? ...and...the flow-tokens in [1] are what make it possible to skip bogus Contact: header and route to an existing flow? Regards Shih-Ping Chan

2 1

KEMI: getting ksr_request_route not found or is not callable after t_relay()
by Nigel Daniels 16 Jul '22

16 Jul '22

``` def ksr_route_relay(self, msg): if KSR.is_method_in("IBSU"): if KSR.tm.t_is_set("branch_route") < 0: KSR.tm.t_on_branch("ksr_branch_manage") if KSR.is_method_in("ISU"): if KSR.tm.t_is_set("onreply_route") < 0: KSR.tm.t_on_reply("ksr_onreply_manage") if KSR.is_INVITE(): KSR.info("try and relay the traffic\n") if KSR.tm.t_is_set("ksr_failure_route") < 0: KSR.tm.t_on_failure("ksr_failure_manage") t_relay_result = KSR.tm.t_relay(["10.0.0.6", "5060"]) KSR.info(f't_relay: {t_relay_result}\n') KSR.info(f'MSG: {msg}') if KSR.tm.t_is_set("ksr_failure_route_one"): KSR.tm.t_on_reply("ksr_on_reply_route_one") ``` ERROR: app_python3 [python_exec.c:91]: apy_exec(): ksr_request_route not found or is not callable Again this only happens after t_replay is called. Any help understanding why this happens would be appreciated. -- - Nigel

2 1

uacreg get error on 2nd 401: error uac_reg_tm_callback(): authentication failed
by Алексей Якимкин 15 Jul '22

15 Jul '22

Hello, I try to register on provider`s pbx version: kamailio 5.7.0-dev0 (x86_64/linux) modparam("uac", "reg_db_url", "mysql://kamailio:123@localhost/kamailio") modparam("uac", "reg_contact_addr", "1.1.1.1:5060") modparam("uac", "reg_retry_interval", 120); # for failed registrations modparam("uac", "reg_keep_callid", 1) modparam("uac","restore_mode","auto") uacreg table: +----+---------+------------+-----------+----------------+-----------------+-------+----------------+------------------+----------+---------------------+---------+-------+-----------+--------------+------------------+ | id | l_uuid | l_username | l_domain | r_username | r_domain | realm | auth_username | auth_password | auth_ha1 | auth_proxy | expires | flags | reg_delay | contact_addr | socket | +----+---------+------------+-----------+----------------+-----------------+-------+----------------+------------------+----------+---------------------+---------+-------+-----------+--------------+------------------+ | 1 | 1234567 | pbx | 127.0.0.1 | login | ip.provider.ru | | login | password| | sip:...com | 120 | 0 | 0 | | 10.130.0.23:5060 | I see this two 401 replies [image: изображение.png] 1st REGISTER Via: SIP/2.0/UDP 1.1.1.1:5060 ;branch=z9hG4bK589d.8183f2e6000000000000000000000000.0 To: <sip:login@ip.provider.com> From: <sip:login@ip.provider.com >;tag=f9cf84029bbf24b61836c3d9fef7b7ae-78a2ffe1 CSeq: 10 REGISTER Call-ID: 6818f1b436befce1-21980(a)10.130.0.23 Max-Forwards: 70 Content-Length: 0 User-Agent: server Contact: <sip:1234567@1.1.1.1:5060> Expires: 120 1st 401 WWW-Authenticate: Digest algorithm=MD5,realm=".......org",nonce=" *56806a6f32c49a3afea043c2fa42381f*",domain="sip:......org",qop="auth", *stale=FALSE* 2nd REGISTER Authorization: Digest username="login", realm=".....", nonce=" *56806a6f32c49a3afea043c2fa42381f*", uri="sip:......com", qop=auth, nc=00000001, cnonce="357420659", response="6e300c9f7f4d81095f29b5408f8d5191", algorithm=MD5 2nd 401 WWW-Authenticate: Digest algorithm=MD5,realm=".....org",nonce=" *3b9586f6d31a76e509eb204d8a030415*",domain="sip:.....org",qop="auth", *stale=TRUE* I think kamailio need to sent 3nd REGISTER with a new nonce, because stale=true. But it interrupts register with error: Jul 14 18:41:00 opensips kamailio[21936]: 10(21949) DEBUG: {2 11 REGISTER 6818f1b436befcf7-21980(a)10.130.0.23} uac [uac_reg.c:852]: uac_reg_tm_callback(): completed with status 401 [uuid: 1234567] Jul 14 18:41:00 opensips kamailio[21936]: 10(21949) ERROR: {2 11 REGISTER 6818f1b436befcf7-21980(a)10.130.0.23} uac [uac_reg.c:933]: uac_reg_tm_callback(): authentication failed for <1234567> Can I get kamailio to send REGISTER after the second 401-reply? Can I intercept this REGISTER and 401 in kamailio.conf? Is there any examples? I know about this instruction https://kamailio.org/docs/modules/5.7.x/modules/uac.html :) Itried in request_route xlog("L_NOTICE", "uac 114411 1 remote fU $fU user [$rU] on [$rd] via [$du] $Ri \n"); if (uac_reg_request_to("$fU", 0)) { xlog("L_NOTICE", "uac 114411 Found remote user [$rU] on [$rd] via [$du]\n"); t_on_failure("REMOTE_AUTH"); #t_relay(); } And failure_route failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "uac 114411 Remote asked for authentication\n"); uac_auth(); } } but I haven't got logs on register request, 401 reply. -- Best regards, Alex

1 1

usage of {s.urldecode.param} to convert special chars (why is + removed?)
by Benoit Panizzon 14 Jul '22

14 Jul '22

Hi List Service Numbers, like to activate CFW probably often start with * and end with a # # is a special character that needs to be urlencoded in an SIP URI. So best would be to use a generic kamailio urldecode transormation. I guess {s.urldecode.param} is what I was looking for. So when I get a call to the 'enable CFW' service I get $rU => "*21+419999999%23" So I want to extract the destination number and remove the urlencoded # sign. So I do $var(cfw-dest) = $var(cfw-dest) = $rU; $var(cfw-dest) = $(var(cfw-dest){s.substr,3,0}); # "+419999999%23" $var(cfw-dest) = $(var(cfw-dest){s.urldecode.param}); But now I end up with 419999999# the + sigh disappeared. Is this expected behavior? How would this be done the correct way? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Releasing v5.6.1
by Daniel-Constantin Mierla 14 Jul '22

14 Jul '22

Hello, I am considering to release Kamailio v5.6.1 (out of branch 5.6) on Wednesday, July 6, 2022 -- in about 1 week and a half from now. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online: June 20-23, 2022 * https://www.asipto.com/sw/kamailio-advanced-training-online/

5 6

Need Suggestion/Idea to set call limit on 3 kamailios.
by Waqar 40 13 Jul '22

13 Jul '22

Please Read the Scenario in full and then guide me. In my scenario, I have three Kamailio (KamA, KamB, KamC) running in production. They do load balancing for the asterisks which are about 9. Kamailios randomly receive calls from the SIP trunk. I have set the limit of 750 calls on each kamailios. I want to provide only 1000 Calls to my client but they are enjoying calls more than this since each kamailio can receive 750 calls (I have set this limit for redundancy purpose). And I cannot set the limit like 750 on KamA and the rest on KamB and KamC for redundancy purposes. If one kamailio goes down then they send calls on the other kamailios so that no call drops. So, I want to know if there is a module or a way that my redundancy is not affected and I limit total calls on the three kamailios to not exceed 1000. Waiting for a prompt response. Let me know if you need any further information. Regards, Vicky

3 3

How to disable GRUU?
by Алексей Дырявко 11 Jul '22

11 Jul '22

Hello. kamailio -v version: kamailio 5.4.8 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 How can I disable GRUU support completely? Sip client Grandstream NT-XXX sends default registration request with "reg-id" and "instance" REGISTER sip:kamailio3.telecom.ru SIP/2.0 Via: SIP/2.0/UDP 172.17.0.252:5060;branch=z9hG4bK1645839794;rport From: "7777" sip:7777@kamailio3.telecom.ru;tag=53756161 To: sip:7777@kamailio3.telecom.ru Call-ID: 308071885-5060-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5060 ;reg-id=1;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB" Authorization: Digest username="7777", realm="kamailio3.telecom.ru", nonce="Ysv842LL+7dABGNcyuYZhl1L2QTPgzFq", uri="sip:kamailio3.telecom.ru", response="0ef43a754a7422dd9d8ed36f91aa546 , algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream HT-502 V2.0A 1.0.9.1 chip V2.2 Supported: path Expires: 1800 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.0.252:5060 ;branch=z9hG4bK1645839794;rport=5060;received=172.17.0.252 From: "7777" sip:7777@kamailio3.telecom.ru;tag=53756161 To: sip:7777@kamailio3.telecom.ru ;tag=5bf75eb19869d3e257cd1fde2cbf92c4.7224507f Call-ID: 308071885-5060-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5060;expires=1800;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=1, sip:07777@172.17.0.250:5060;expires=101, <sip:7777@172.1 0.250:5061>;expires=101, sip:7777@172.17.0.252:5062 ;expires=607;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=2 Server: kamailio (5.4.8 (x86_64/linux)) Content-Length: 0 REGISTER sip:kamailio3.telecom.ru SIP/2.0 Via: SIP/2.0/UDP 172.17.0.252:5062;branch=z9hG4bK212965390;rport From: "7777" sip:7777@kamailio3.telecom.ru;tag=1540711120 To: sip:7777@kamailio3.telecom.ru Call-ID: 1127299103-5062-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5062 ;reg-id=2;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB" Authorization: Digest username="7777", realm="kamailio3.telecom.ru", nonce="Ysv862LL+78cGZE/vXJnQfemhCMiBilm", uri="sip:kamailio3.telecom.ru", response="6c3333159d41ae0fcaabfbbd8c5026e , algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream HT-502 V2.0A 1.0.9.1 chip V2.2 Supported: path Expires: 1800 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.0.252:5062 ;branch=z9hG4bK212965390;rport=5062;received=172.17.0.252 From: "7777" sip:7777@kamailio3.telecom.ru;tag=1540711120 To: sip:7777@kamailio3.telecom.ru ;tag=5bf75eb19869d3e257cd1fde2cbf92c4.451b5a53 Call-ID: 1127299103-5062-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5060;expires=1792;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=1, sip:7777@172.17.0.250:5060;expires=93, <sip:7777@172.17 .250:5061>;expires=93, sip:7777@172.17.0.252:5062 ;expires=1800;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=2 Server: kamailio (5.4.8 (x86_64/linux)) Content-Length: 0 Later, when I use branching (with q=-1.0), the kamailio sends an Invite to only one Contact from 7777@172.17.0.252:5062 If I use another Sip client without "instance" , such as Linksys-PAP2T, branching works fine. GRUU is always disabled modparam("registrar", "gruu_enabled", 0)

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(Off-topic) Vienna, meeting
by Henning Westerholt 09 Jul '22

09 Jul '22

(Off-topic) Hello, In about two weeks I will be a few days in Vienna (12h - 14th July). If you want to meet for a drink to chat about Kamailio and other topics, just contact me off-list. Thanks, and regards, Henning

1 1

Client behind nat without stun and kamailio behind NAT
by Алексей Якимкин 08 Jul '22

08 Jul '22

Hello, I hope somebody could help me. This is my scheme. User-agent is behind NAT1. Kamailio and pbx are behind NAT2 [Client ip-phone 192.168.89.213 without stun] - LAN1 - NAT1(46.0.0.30) - (internet) - (51.0.0.60)NAT2 - Local2 - (10.130.0.23:5060)kamailio( 10.130.0.23:5070) - pbx *Questions*: *1*. The VIA header (with 10.130.0.23 and 51.0.0.60) wasn't included in SIP packets. Why? For example 200OK reply. It came from pbx through kamailio. Which setting could break it? Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK2480172053 Record-Route: <sip:c172.19.19.111.8348.call.cgatepro;lr> Record-Route: <sip:172.19.19.111:5060;lr> Record-Route: <sip:10.130.0.23:5070 ;r2=on;lr;ftag=4268683942;nat=yes>,<sip:51.0.0.60;r2=on;lr;ftag=4268683942;nat=yes> From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=4268683942 To: <sip:2961@mail.domain.ru:5060 >;tag=A03E2397-404246-FA7543E4_jizmelr-582D Call-ID: 6_1903330087(a)192.168.89.213 [Generated Call-ID: 6_1903330087(a)192.168.89.213] CSeq: 2 INVITE Contact: <sip:signode-404246-FA7543E4_jizmelr-582D@172.19.19.111 ;alias=51.0.0.60~5060~1> Supported: 100rel,timer,replaces,histinfo,precondition Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,INFO,MESSAGE,PRACK,UPDATE,REFER Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 1170 Message Body *2*. About Registrar, Path and $du. Phone set Register with headers: Via: SIP/2.0/UDP 192.168.89.213:5060;branch=z9hG4bK2691182696 From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=3926879477 To: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060> Contact: <sip:a.yakimkin@192.168.89.213:5060> Kamailio respond Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK617463686 Path: <sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on>,<sip:51.0.0.60;lr;received=46.0.0.30~9570~1;r2=on> From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=3926879477 To: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=194ED16D Contact: <sip:a.yakimkin@192.168.89.213:5060>;expires=360 Contact: <sip:2447@192.168.89.221:5060>;expires=247 Contact: <sip:2447@192.168.9.16:5060>;expires=2116 I try to make a call from pbx to ip phone. pbx inserts in Invite header Route: <sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on> But kamailio relayed Invite direct to 192.168.89.213. (There is network connectivity between ip-phone and kamailio through vpn). The code below helps me to solve my issue. I saw mail-list with similar trouble. But no setting could get kamailio to relay Invite to "Route-received" ip. $var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); } *Kamailio settings:* I have such listeners listen=udp:10.130.0.23:5070 # to local network listen=udp:10.130.0.23:5060 advertise 51.0.0.60:5060 # to internet #MODULE SETTING #--- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("path", "use_received", 1) modparam("path", "enable_r2", 1) modparam("path", "received_format", 1) # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "auto_inv_100_reason", "Trying") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 2) modparam("rr", "force_send_socket", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) modparam("registrar", "use_path", 1) modparam("registrar", "path_use_received", 1) modparam("registrar", "path_mode", 0) #--- For register I use this code route[REGISTRAR] { ... add_path_received(); set_send_socket("udp:10.130.0.23:5070"); route(DISPATCH); ... } route[RELAY] { ... if ($Ru eq "sip:10.130.0.23:5070") { $fs = "udp:10.130.0.23:5060"; } else { $fs = "udp:10.130.0.23:5070"; } ... } route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #send INVITE to IP:PROT from Route:...;received= $var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); } if (client_nat_test("3")) { if(nat_uac_test("18")) { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("co", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("co"); } if (is_method("REGISTER")) { #if ($Ru == "sip:10.130.0.23:5070") { # fix_nated_contact(); #} else { # set_contact_alias(); #} set_contact_alias(); } else { if(is_first_hop()) { set_contact_alias(); } else { add_contact_alias("51.0.0.60", "5060", "udp"); #fix_nated_contact(); } } } else { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("cor", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("cor"); } } } else { rtpproxy_manage("co"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); #fix_contact(); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); #} else { #fix_contact(); #} } } #!endif return; } Thank you. -- Best regards, Alex

2 1

log any rpc command
by Patrick Karton 08 Jul '22

08 Jul '22

hello is there a way to log in kamailio any kamcmd entered ? Thanks.

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