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42 participants
37517 discussions

How to externally trigger htable cache flushing?
by Benoit Panizzon 18 Jul '22

18 Jul '22

Hi List We cache stuff like call forwarding information in a htable cache to minimizie SQL lookups and have it available via DMQ on all instances. But of course, if the call forwarding information for a destination changes in the database, we would like to flush either the whole cache, or even better, that single entry. What is the best way to do this? 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 ______________________________________________________

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dialog timeout_avp no longer works?
by Alex Balashov 18 Jul '22

18 Jul '22

Hi, I discovered recently that, on Kamailio 5.5.4 at least, the dialog ’timeout_avp’ no longer appears to have effect. Here are my (seemingly relevant) dialog parameters: modparam("dialog", "enable_stats", 0) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "dlgcor") # DLG "correlator" modparam("dialog", "dlg_flag", 2) modparam("dialog", "default_timeout", 28800) # 8 hours modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "table_name", "dialog") modparam("dialog", "send_bye", 1) modparam("dialog", "wait_ack", 0) modparam(“dialog”, “timeout_avp”, “$avp(dlg_timeout)”) The flow is roughly like this: request_route { ... t_check_trans(); if(is_method(“INVITE”)) { ... dlg_manage(); $avp(dlg_timeout) = 10; ... if(!t_relay()) sl_reply_error(); return; } I was able to resolve the problem by switching to dlg_set_timeout() instead, but I wonder if there has been a regression. Last version I have on which this positively works is 4.4.7, though, so not sure when it might have broken. One nuance: $avp(dlg_timeout) is not initialised prior to calling dlg_manage(), and the value is only set if certain conditions are true. Not sure if some stricter check appeared in later releases which could cause this to be a problem. Thank you! — Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Nathelper sip ping and Path support ?
by matteo brancaleoni 18 Jul '22

18 Jul '22

Hello, I'm trying to setup the following scenario with sip ping: UAC -> Kamailio Edge -> Kamailio Registrar Edge proxy is a multihomed proxy with public/private ips and the registrar is a kamailio on a private network. The edge acts as a nat proxy and loadbalancer. Edge proxy inserts a Path header (and contact alias) which get correctly saved on registrar usrloc. Registration works as expected. Now, since I need to keep nat open towards the uacs and expire them when they're not reachable anymore, I'm trying to send sip pings from the registrar via the edge proxy. I'm using nathelper sipping in the registrar. What I see is that the registrar sends out a sip options putting in the Route header the Path as expected, but the sip packet gets sent directly to the UAC contact and not via the edge, as the Route header specifies. After looking into the nathelper code, it seems that the sip packet is sent via udp directly without any specific routing. So the question: it should work what I'm trying to achieve? Maybe not? Probably I can work around it setting the ;received in the contact on registrar (it will set to the edge proxy) and since the contact is aliased on the edge when the edge receives an option from the registrar it can process the alias and send it to the correct endpoint, but seems a workaround? Or using the keepalive module maybe, even if that means setting specific timers and manually do what the nathelper module already does? Any hint? Thanks, Matteo

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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

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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

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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

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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/

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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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