- sr-users - kamailio.org

kamailio rtp connection
by Pieter Muller 28 Sep '18

28 Sep '18

Good day all I hope Im on the right place to ask the question. Im a newbie to kamailio and have a little problem understanding how to set kamailio correctly. Scenario: VoIP Supplier (IP:161.161.252.20{Private Net}) à Connect via routing to Kamailio Server(155.155.16.2 eno2{private Net}),Kamailio Server(187.221.197.252 eno1{public facing})àconnect to PBX(public IP set in dispatcher) à connect to RTP server(Public IP) Now the problem is that if you call in Number(a)161.161.252.20 <mailto:Number@161.161.252.20> is send to kamailio, kamailio then sends it down to dispatcher IP and that PBX responds directly back via rtp server to 161.161.252.20. The 161.161.252.20 IP is private and cannot be reached from the cloud and vice versa . How do I get kamailio to be the middle man between 161.161.252.20 and RTP server(all traffic must go via the kamailio server) Is there specific modules I must load. Im running Kamailio 4.4 Any help would be muchly appreciated. Regards Pieter

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Important: new module exports interface
by Daniel-Constantin Mierla 28 Sep '18

28 Sep '18

Hello, as a result of collaborative work at Kamailio Developers Meeting, we succeeded to merge the two existing module exports interface (one for kamailio modules and the other one from ser modules) in a single one. All public modules were updated, but if you have any private module then you have to update as well in order to get it compiled with the latest master. Just look at one of the modules (e.g., sl is a good option) and all the fields in mod exports structure have comments with their meaning. In short: we removed unused fields for statistics, mi commands and extra-processes (from kamailio old interface) and oncancel (from ser old interface), kept the rpc exports from ser interface and pv exports and dlopen flags from kamailio interface. The other fields were common in both interfaces, but be aware that order was also changed. If you have a module implementing the old ser interface, you also need to add the free_fixup_field in functions exports tructure -- you can just set it to 0. With this we have now a single mod exports interface in all modules, the core is also cleaner as we got rid of the various compatibility layers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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INFO: maintenance work on deb.kamailio.org
by Daniel-Constantin Mierla 28 Sep '18

28 Sep '18

Hello, we will do some maintenance work on deb.kamailio.org server these days in order to move to a better infrastructure, so it may not be available for a while -- the DNS needs to be updated as well, this also can affect you if your router/provider does caching. Once we finish with the migration, we will send updates. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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Re: [SR-Users] No dispatcher failover
by Gertjan Wolzak 28 Sep '18

28 Sep '18

Alex, Thank you for the feedback, just the info I needed. Failover is working now just as it should. Now I just have to tweak the failover timing. Rgds, Gertjan Wolzak

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Cannot handle Cancel Request properly
by Aqs Younas 28 Sep '18

28 Sep '18

Greetings list, I am getting hard time to make kamailio perfectly handle cancels request. I am using http_async_client and MSILO to handle push notifications. Case 1. When B party record exists in location but it is off-line and I send cancel *instantly* from Party A while kamailio is sending push to wake B Party up. modparam("tm", "failure_reply_mode", 3) modparam("tm", "cancel_b_method", 1) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) You can see in below picture 407 being generated after some time. [image: default.PNG] What could be the reason of this problem? any pointer is much appreciated. Br, Aqs.

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replay of tls delayed by 30 secs.
by Andrew Chen 28 Sep '18

28 Sep '18

Hi Kam community, Facing an urgent issue here and not sure how to troubleshoot it further. So our engineers been getting reports of unknown SIP timeouts and they suspect it's something on the Kamailio side. I've logged every transaction from receiving the message to done relaying it as shown below: Sep 24 18:38:13 sjoprodfnsg01 /usr/local/sbin/kamailio[42608]: INFO: <script>: [1589Fz-ses1-41d9a88aR70125425(a)xxx.xxx.xxx.xxx][4534123][CSeq 7] NG-FuzeMedia-Server/42ae58c/mediahub Relaying [UPDATE] from (xxx.xxx.xxx.xxx ) to ( xxx.xxx.xxx.xxx ) via (tls) Sep 24 18:38:13 sjoprodfnsg01 /usr/local/sbin/kamailio[42608]: INFO: <script>: [1589Fz-ses1-41d9a88aR70125425@ xxx.xxx.xxx.xxx ][4534123][CSeq 7] NG-FuzeMedia-Server/42ae58c/mediahub Done relay [UPDATE] from (xxx.xxx.xxx.xxx ) to ( xxx.xxx.xxx.xxx ) via (tls) When I pulled the 443 SIP trace from my tcpdump session, it shows the application data left the machine at 18:44 as shown here: [image: image.png] Note: This packet is encrypted and can't see the actual SIP message. This matches what was received on the client logs: 2018-09-24T18:38:44.032Z [40927] [INF] fuze.sip SipParser.cpp:549 LogSipMessage: xxx.xxx.xxx.xxx:443 - 4292B (local time 09/24 11:38:44) -------------------- RECEIVE: UPDATE sips:Luke_Surazski@ xxx.xxx.xxx.xxx:50795;id=37262838;inst=52241254 SIP/2.0 Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:443;branch=z9hG4bK15f.5052c318e1766a0a9405f10708f43188.0;i=acbb7 Via: SIP/2.0/TCP xxx.xxx.xxx.xxx :38342;rport=38342;branch=z9hG4bK-55f913-t9-I214262897 What would cause such delays? I'm running Kamailio 4.1.5 Thanks. -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) 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.*

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subst_hf to rewrite a contact header
by Ben Hood 27 Sep '18

27 Sep '18

Hi, I’m wondering how to use subst_hf() properly. I’m trying to debug a broken flow by manually rewriting the contact header from Contact: <sip:b73c6f29-0101-4802-afcd-efb63f1e6d8f@10.10.10.12:5090;transport=udp> to Contact: <sip:2018092417381900003@81.x.x.x> by using $var(ctct) = "<sip:2018092417381900003@81.x.x.x>"; subst_hf("Contact", “/\<.+\>/$var(ctct)\r\n/", "a”); but the result is Contact: <sip:2018092417381900003@81.x.x.x> sip:b73c6f29-0101-4802-afcd-efb63f1e6d8f@10.10.10.12:5090;transport=udp Testing the regex with an external tool appears to match all of the original header, so I’m wondering why subst_hf doesn’t appear to rewrite the entire header. Is there something I’m doing wrong? TIA, Ben

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Documentation of Kamailio.cfg
by Prabhat Kumar 27 Sep '18

27 Sep '18

Hi everyone, I there any suitable documentation for understating kamailio configuration file. Regards, Prabhat Kumar Sr. Software Developer [image: phone] +91 92129 92129 Ext: 89 [image: moible] +91 80831 10443 [image: Facebook] <https://app.mailtag.io/link-event-v2?mt__id=379f500e-c250-11e8-a3de-0625822…>[image: Twitter] <https://app.mailtag.io/link-event-v2?mt__id=379f5248-c250-11e8-9d85-0625822…>[image: Twitter] <https://app.mailtag.io/link-event-v2?mt__id=379f5392-c250-11e8-b97b-0625822…>[image: Linkedin] <https://app.mailtag.io/link-event-v2?mt__id=379f54be-c250-11e8-8e23-0625822…>[image: Google+] <https://app.mailtag.io/link-event-v2?mt__id=379f55f4-c250-11e8-affd-0625822…> [image: Logo] <https://app.mailtag.io/link-event-v2?mt__id=379f570c-c250-11e8-a472-0625822…>

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Kamailio - Asterisk integration
by denislav.tsonev＠musala.com 27 Sep '18

27 Sep '18

Hello guys, I'm following this guide: https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And I have the following schema: Phone A -----> Kamailio -----> Asterisk -----> Phone B (10.0.8.249) (10.0.6.123) (10.0.3.144) (10.0.13.23) I think the flow it is like (because when UAC register to Asterisk, the contact points to Kamailio IP): 1. Phone A --> Kamailio --> Asterisk 2. Asterisk --> Kamailio --> Phone B In attachment is the pcap of the call, kamailio config. The problem is that I never receive ACK after 200 OK and call drops after 30 seconds? Kamailio version 5.1 and Asterisk version 15.1 Phones are grandstreams and also tested with microsip soft client and same results... Like "Phone A" never sends ACK back I tried different approaches to change contact header in 200 OK, because is Asterisk IP. Disable NAT handling. Different "nat_uac_test" flags, but still without success :( Also tried with this NAT detection and manage routes (again without any success) - source https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal 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; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } Thanks in advance Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

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Problem with replicating dialogs using DMQ
by jenus＠cyberchaos.nl 27 Sep '18

27 Sep '18

Hello, I'm trying to replicate active dialogs using the DMQ module. I see the dialogs on the other nodes but it generates a error message in the logs on all the nodes: Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_timer.c:130]: insert_dlg_timer(): Trying to insert a bogus dlg tl=0x7efdca1bebf8 tl->next=0x7efdca14cb80 tl->prev=0x7efdca14cb80 Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_dmq.c:309]: dlg_dmq_handle_msg(): Unable to insert dlg timer 0x7efdca1beb98 [2643:3725] Even with the timeout_avp option disabled i still see the error messages. I tested this on kamailio 5.0.3 and 5.1.3 with the same results. With debug enabled i see the correct data is replicated to the other nodes: dlg_dmq_replicate_action(): sending serialized data {"action":2,"h_entry":1328,"h_id":409,"state":4,"start_ts":1538036606,"lifetime":7200,"tag1":"1635359980","tag2":"as2d80799c","cseq1":"21","cseq2":"0","route_set1":"","route_set2":"","contact1":"sip:sipuser@10.10.10.10:1057;alias=10.10.10.10~1055~1","contact2":"sip:+31101234567@10.10.10.10:5060"} "lifetime":7200 is in the serialized data, and when i check on the other node with "kamcmd dlg.list" i see that the lifetime data is not updated. Thanks, Jan

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create domain "on the fly" in registrar modukle
by Maarten Ureel 27 Sep '18

27 Sep '18

I'm building a setup with FreeSWITCH and Kamailio 5.2 but I'm having troubles in handling the registrations for multiple domains. So far I have in my config these related settings: # Disable database completely modparam("usrloc", "db_mode", 0) # Take domain into account when saving modparam("usrloc", "use_domain", 1) # Preload location table modparam("usrloc", "preload", "location") I'm using the app_python module; when performing the registrar save like this: KSR.registrar.save(KSR.pv.get('$fd'), 1) It fails, the debug shows that it cannot find the domain that the SIP user has here. When I use: KSR.registrar.save('location', 1) The phone registers just fine; however I'm bothered by the fact that the documentation states that you should specify the domain as a parameter. Is it OK if everything is in one 'location' domain? How can I let Kamailio know what domains exist (where does Kamailio get this information from)? We have a unique domain for every customer, like g123.ctx.ourcompany.tld. Kind regards Maarten Ureel

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Adding new rtpproxy instance without restarting Kamailio
by Pintu Lohar 27 Sep '18

27 Sep '18

Hi All, I have a very basic question. Is there any way to add new rtpproxy instance without restarting Kamailio? As per my understanding, we can reload the dispatcher table when Kamailio server is running with kamctl dispatcher reload but do we have any such for rtpproxy list also from db when rtpproxy table ? Below is my rtpproxy related config, load from DB , #Once RTPProxy was found unreachable and marked as disabled, #the rtpproxy module will not attempt to establish communication to RTPProxy for rtpproxy_disable_tout seconds. modparam("rtpproxy", "rtpproxy_disable_tout", 120) #only for local host. modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2") modparam("rtpproxy", "db_url", "mysql://kamailio:xxxxxxxx@10.xx.xx.xx /kamailio") modparam("rtpproxy", "table_name", "rtpproxy") #enable marker for sdp modification by rtpproxy modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n") my current instance : ./kamcmd rtpproxy.list { setid: 0 url: udp:xxx.xx.xxx.xxx:7722 index: 2 disabled: 0 weight: 1 recheck: 0 } Db +----+-------+--------------------------+-------+--------+-------------+ | id | setid | url | flags | weight | description | +----+-------+--------------------------+-------+--------+-------------+ | 1 | 0 | udp:XXX.XXX.XXX.XXX:7722 | 0 | 1 | | +----+-------+--------------------------+-------+--------+-------------+ I already explored ./kamcmd rtpproxy.enable command , but it seems that this command is useful for re-enabling or disabling existing rtpproxy instance . My purpose is to add new rtpproxy instance when Kamailio is running. Thanks in advance. Any suggestion would very helpful. Thanks & Regards Pintu

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KAMAILIO 5.1 : How to enable and subscribe to all of the SIP events and kamailio server through tcp, json
by yboujraf＠by-systems.be 26 Sep '18

26 Sep '18

Dear All, I am currently working to control from third party system Kamailio I have there are several way to interact with Kamailio but not sure I could make what I am expecting My question is : * I would like to get from my third party system subscribe to all of the sip events and status of the kamailio server My best way is TCP, JSON, scrambled if possible It will be great if anybody could route me to the correct direction. Best Regards, Youssef --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

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KAMAILIO 5.1 - how to make a full redundancy of two kamailio Proxy server ?
by yboujraf＠by-systems.be 26 Sep '18

26 Sep '18

Dear All, I am currently thinking to use TWO (2) Kamailio Proxy server in redundancy. The system is : * I have one (1) public address * Two (2) Kamailio Servers with two different IP address My questions are : * What I need to add in front of the two Kamailio servers to be sure sip phone will be connected through the public ip address to one of the kamailio server at time. I have seen we could enable "cluster module" to sync the dabase of the two (2) kamailio server at the same time but not for the request. * Script to add to be sure the kamailio servers are correctly configured. Best Regards, Youssef --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

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kamailio not forwarding 407
by David Villasmil 26 Sep '18

26 Sep '18

Hello guys, I have an almost default config, the box behind kamailio sends out the register, it is forwarded to the provider, which replies with 401, this gets forwarded back to the box which sends out the REGISTER with credentials and all works well. BUT when calling out, INVITE is sent to kamailio which is properly forwarded to the provider. The provider replies with a 407 and this 407 is NOT forwarded back to the box. Any ideas why? I can't figure it out! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Dispatcher's Probing
by Duarte Rocha 26 Sep '18

26 Sep '18

Hey, I've been using the Dispatcher module with great results. But now, i want to be able to have some gateways to belong to more than one group set. To achieve that, i'm duplicating the Gateway in the Dispatcher's table and give it a different group set. However, if i do this, Dispatcher will do repeated probing to this gateways. Is there any way to avoid this or to achieve the same result in a different way? Best Regards

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No dispatcher failover
by Gertjan Wolzak 26 Sep '18

26 Sep '18

Hello Kamailions, I want to send traffic to an Asterisk box but when that box fails traffic should be routed to the second Asterisk box. The dispatcher params are as follows: # ----- Dispatcher params ------ modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") modparam("dispatcher", "ds_inactive_threshold", 4) modparam("dispatcher", "ds_ping_latency_stats", 1) modparam("dispatcher", "ds_probing_threshold", 1) My dispatcher.list is: #DBBS PBX # SBC's 1 sip:ip_address_sbc:5060 0 0 weight=100 #PBX 2 sip:ip_pbx_01:5060 0 10 weight=100 2 sip:ip_pbx_02:5060 0 5 weight=100 Verry simple request route: request_route { if (is_method("OPTIONS")) { sl_send_reply("200", "OK"); exit; } #Call from SBC to PBX #Get gateway from dispatcher list if($si=="ip_address_sbc1" |$si=="ip_address_sbc2" ) { ds_select_dst("2", "8"); $var(new_uri) = "sip:" + $tU + "@" + $dd + ":5060"; $ru=($var(new_uri)); route(RELAY); exit; } # Call from pbx to SBC # Get sbc ip from dispatcher list ds_select_dst("1","4"); $var(new_uri) = "sip:" + $tU + "@" + $dd + ":5060"; $ru=($var(new_uri)); route(RELAY); } When I send a call to the kamailio its routed correctly to the first pbx. But when I kill the asterisk process on the first pbx, an invite is still send by Kamailio to the first pbx. Maybe I am assuming incorrectly that when the dispatcher does not react to the ds_ping it will be set to "inactive" and not be selected by the ds_select_dst command... Or am I thinking to easy and do I need to configure some actions in the failure route? Pretty sure I'm doing something wrong, but cant see where and have not been able to find the information. Tips are appreciated. Rgds, Gertjan Wolzak

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onsend_route showing wrong answer after forking
by Aymeric Moizard 26 Sep '18

26 Sep '18

Hi, I'm experimenting "onsend_route" with "onsend_route_reply=yes" to improve my logs. I was surprised to see that "onsend_route" doesn't always display the "forwarded" answer after negative final answers. The documentation seems to indicate that this route will execute only if "the received reply is sent out". -as I'm forking, the final 6xx answer is not "forwarded" immediatly... and I guess this is the reason it doesn't show. Here comes a real strange use-case: With forking enabled, only the last answer will appear in the "onsend_route", however, in some case, this is NOT the answer being forwarded. Example: 1/ receive 603 decline from branch1 2/ a CANCEL is sent on branch2 3/ a 487 Request Terminated is received on branch2 What happens: 1/ onsend_route is showing "487 Request Terminated" 2/ 603 Decline is forwarded (as expected) The documenation say: "If set to 1 (yes, on), onsend_route block is executed for received replies that are sent out. Default is 0." This doesn't look to be correct. (either the doc, or the code?) Is there any possibility to install a log in the script that would log ALL *exact* replies being sent: 1/ the real forwarded ones 2/ and also the ones sent by kamailio directly. May be this is possible with some "event_route"? Tks a lot! Aymeric -- Antisip - http://www.antisip.com

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kamailio 5.0.6 | segfault bad msg or missing FROM header
by Laurent Schweizer 26 Sep '18

26 Sep '18

Dear all, I have setup few weeks ago a kamailio 5.0.6, was working correctly till last night where 2 time he stop with a segfault error : Sep 25 23:36:19 ch0012 kamailio[18696]: ERROR: <core> [core/parser/parse_from.c:53]: parse_from_header(): bad msg or missing FROM header Sep 25 23:36:19 ch0012 kernel: kamailio[18696]: segfault at 0 ip 00000000004d7c4f sp 00007fff16135130 error 4 in kamailio[400000+42b000] Sep 25 23:36:19 ch0012 kernel: kamailio[18682]: segfault at 0 ip 00000000004d7c4f sp 00007fff16135130 error 4 in kamailio[400000+42b000] Sep 25 23:36:19 ch0012 kamailio[18696]: ERROR: siptrace [siptrace.c:535]: sip_trace_prepare(): cannot parse FROM header Sep 25 23:36:19 ch0012 kamailio[18682]: ERROR: <core> [core/parser/parse_from.c:53]: parse_from_header(): bad msg or missing FROM header Sep 25 23:36:19 ch0012 kamailio[18682]: ERROR: siptrace [siptrace.c:535]: sip_trace_prepare(): cannot parse FROM header Sep 25 23:36:19 ch0012 abrt-hook-ccpp: Process 18696 (kamailio) of user 0 killed by SIGSEGV - ignoring (repeated crash) Sep 25 23:36:19 ch0012 abrt-hook-ccpp: Process 18682 (kamailio) of user 0 killed by SIGSEGV - dumping core Sep 25 23:36:20 ch0012 abrt-hook-ccpp: Saved core dump of pid 18696 to core.18696 at / (2294079488 bytes) Sep 25 23:36:20 ch0012 kamailio[18759]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 31 Sep 25 23:36:20 ch0012 kamailio[18679]: ALERT: <core> [main.c:743]: handle_sigs(): child process 18696 exited by a signal 11 Sep 25 23:36:20 ch0012 kamailio[18679]: ALERT: <core> [main.c:746]: handle_sigs(): core was generated Sep 25 23:36:20 ch0012 kamailio[18679]: INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD Sep 25 23:36:20 ch0012 kamailio[18759]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18754]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18758]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18757]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18753]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18752]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18756]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18751]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18746]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18747]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received Sep 25 23:36:20 ch0012 kamailio[18750]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received I have exported from the core file some info (file is attached), any idea ? BR Laurent

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Dispatcher Questions
by Duarte Rocha 26 Sep '18

26 Sep '18

Greetings, I have some questions about Dispatcher0s modules general behaviours. Can Dispatcher use two tables to read values from? If I can't use two tables, is it possible to read them from a Mysql View? I want to be able to have a Gateway belonging to more than one Gateway Set. By doing so, in the table, the gateway will be repeated but with different groups. How will the probing work in this scenario? Will the Gateway be probed multiple times? Thanks for your help. Cheers

2 1

TM Module timing question
by Igor Olhovskiy 25 Sep '18

25 Sep '18

Hi! A bit lost in timers for TM module Idea is following I want to have dial timeout for 30 seconds, but only if I received 100 Trying in reply to INVITE. If no 100 Trying is received - I want timeout to be like 2 seconds. 100 Trying is indicator for me, that other side is alive. As I got, fr_timer and fr_inv_timer is not what I’m looking for. Regards, Igor

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htable event routes behavior
by Володимир Іванець 25 Sep '18

25 Sep '18

Hello all, I have following configuration: 1. Htable which loads from MySQL DB and has autoexpire setting. *modparam("htable", "htable", "tl_connect_users=>size=4;autoexpire=120;dbtable=some_htable")* 2. Regular *route[SOME_ROUTE]* with sole logic. 3. *event_route[htable:mod-init]* 4. *event_route[htable:expired: some_htable]* I use while loop in *event_route[htable:mod-init]* to extract stored values from htable, assign them to *$var(some_variable)* and call *SOME_ROUTE* to perform some action for all of them on Kamailio start up. *SOME_ROUTE* has *$sht(some_htable =>$var(some_variable)) = "enabled"* line to add expired value back to the htable. The pboblem is when* event_route[htable:expired: some_htable]* triggers *SOME_ROUTE*. Values are not being added to htable. Hovewer if I will try to manipulate values for key_name or key_value by adding some character new item will be insert to the htable. Does anybody know if behavior is correct when exactly same value can not be inserted back to htable on it's expired event? Thanks a lot!

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Help with kamailio/freeswitch
by David Villasmil 25 Sep '18

25 Sep '18

Hello guys, I know this is kamailio's mailing list, but I'm getting no answer on freeswitch, and i thought someone here has probably done this is the past. I have a proxy in front of freeswitch. I want freeswitch to use the proxy for everything including the termination to my provider. If I configured the gateway like this: <include> <gateway name="myprovider"> <param name="username" value="username" /> <param name="auth-username" value="username"/> <param name="password" value="password" /> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="from-domain" value="myprovider.com"/> <param name="realm" value="myprovider.com"/> <param name="register-proxy" value="myproxy.domain.com"/> </gateway> </include> The registration works properly, the REGISTER goes to the proxy like REGISTER sip:myprovider.com;transport=udp SIP/2.0 and the proxy forwards it fine, and registration works great. But when i try to call via this gateway, the call goes STRAIGHT to the provider! If, on the other hand, i configure my gateway as: <include> <gateway name="myprovider"> <param name="username" value="username" /> <param name="auth-username" value="username"/> <param name="password" value="password" /> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="from-domain" value="myprovider.com"/> <param name="realm" value="myprovider.com"/> <param name="register-proxy" value="myproxy.domain.com"/> <param name="proxy" value="myproxy.domain.com"/> </gateway> </include> Registration does NOT work, because the uri is: REGISTER sip:myproxy.domain.com;transport=udp SIP/2.0 Because there is no registrar on that proxy!

3 2

escape character
by David Villasmil 25 Sep '18

25 Sep '18

Hello guys, I'd like to search for " ' or & But i can't seem to figure out how... I tried: $tU =~ "[\"'&]" But that's not catching it... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Substitution Regex
by Duarte Rocha 24 Sep '18

24 Sep '18

Greetings, I'm using subst_hf() function to do some search and replace and i want to identify a undifined amount of digits after a string (ex: string=123123). Everywhere i look, the right way seems to be \d so i'm doing subst_hf("Via","/;string=\d/substitution_string/", "a"); It doesn't seem to work. Any ideas why? Best Regards, Duarte Rocha

3 2

Issuse with http_client and config_file
by Barry Flanagan 24 Sep '18

24 Sep '18

Hi I am trying to use http_client with a config_file set but can't get it to work .Have tried on 4.4 and 5.1 with the same results (Debian 9 packages). Here are my modparams modparam("http_client", "config_file", "/etc/kamailio/httpconnections.cfg") ..and here is the contents of httpconnections.cfg [apiserver] url = https://my.api.com/v1 username = apiuser password = api_password timeout = 1 authmethod = 2 maxdatasize = 4 tlsversion = TLSv1.2 verify_peer = yes Whenever I try to run this, I get the following error: http_client [curlcon.c:142]: curl_get_connection(): curl_get_connection no success in looking for httpcon: [apiserver] I am running the folowing: $var(res) = http_connect("apiserver", "/wlrmap/$var(wlrUserFromPAI).json", "$avp(wlrdata)"); >From the debug logs it appears that Kamailio is initialising the module OK (included below) so I am not sure why it is failing to find it subsequently. Any help appreciated! Thanks -Barry Flanagan Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: <core> [core/sr_module.c:988]: init_mod(): http_client Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:254]: mod_init(): init curl module Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:799]: curl_init_con(): curl_init_con httpcon: [apiserver] ID 3613147223 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:841]: curl_init_con(): CURL: Added connection [apiserver] Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:722]: fixup_raw_http_client_conn_list(): cname: [apiserver] url: [vapi-de v.op.telinsta.net/operator/v1] username [api%40operator-dev.telinsta.net] password [k3645QA36VtvsSLgo5m9Y827NCeKfPjhRnd879zx] failover [] timeout [1] useragent [kamailio (5.1.5 (x86_64/linux))] maxdatasize [0] Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [curlcon.c:728]: fixup_raw_http_client_conn_list(): cname: [apiserver] client_cert [ (null)] client_key [(null)] ciphersuites [(null)] tlsversion [6] verify_peer [0] verify_host [0] keep_connections [0] Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:297]: mod_init(): **** init http_client module done. Curl version: 7. 52.1 SSL OpenSSL/1.0.2l Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:299]: mod_init(): **** init http_client: Number of connection objects : 1 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:301]: mod_init(): **** init http_client: User Agent: kamailio (5.1.5 (x86_64/linux)) Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:303]: mod_init(): **** init http_client: HTTPredirect: 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:305]: mod_init(): **** init http_client: Client Cert: Key Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:308]: mod_init(): **** init http_client: CA Cert: (null) Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:309]: mod_init(): **** init http_client: Cipher Suites: Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:311]: mod_init(): **** init http_client: SSL Version: 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:312]: mod_init(): **** init http_client: verifypeer: 1 verifyhost: 2 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:314]: mod_init(): **** init http_client: HTTP Proxy: Port 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:317]: mod_init(): **** init http_client: Auth method: 2 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:318]: mod_init(): **** init http_client: Keep Connections open: 0 Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: http_client [http_client.c:321]: mod_init(): **** Extra: Curl supports TLS IPv6 IDN Sep 24 14:01:31 wlr-proxy-01 /usr/sbin/kamailio[31937]: DEBUG: <core> [core/route.c:872]: fix_actions(): fixing http_connect()

1 0

Module RTPPROXY param Timeout_Socket not Working
by Hamid Hashmi 24 Sep '18

24 Sep '18

Kamailio didn't start listening on IP mentioned in Module RTPPROXY parameter Timeout_Socket<https://kamailio.org/docs/modules/5.1.x/modules/rtpproxy.html#rtpproxy.p.ti…>. Below is my configuration file. loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_tout", 1) modparam("rtpproxy", "rtpproxy_retr", 2) modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2") loadmodule "xmlrpc.so" modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "mode", 1) modparam("xmlrpc", "url_match", "^/RPC2") when I type a command $ netstat -nlp it only shows SIP listening ports. Kamailio should also start listening on 127.0.0.1:8000. Below are the RTPproxy logs, RTPproxy is also unable to bridge the media DBUG:get_command:GLOBAL: received command "19083_4 Uc120,101 9a1060af660d4202976c419ab5300470 58.65.176.42 4000 1ea06d1eb0a8443697a08e568045598c;1" INFO:rtpp_command_ul_handle:GLOBAL: new session 9a1060af660d4202976c419ab5300470, tag 1ea06d1eb0a8443697a08e568045598c;1 requested, type strong INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new session on a port 35012 created, tag 1ea06d1eb0a8443697a08e568045598c;1 INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: pre-filling caller's address with 58.65.176.42:4000 DBUG:rtpc_doreply:GLOBAL: sending reply "35012 51.38.201.3 " DBUG:get_command:GLOBAL: received command "19083_5 Uc120,101 9a1060af660d4202976c419ab5300470 58.65.176.42 4022 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 1ea06d1eb0a8443697a08e568045598c;1 xmlrpc:http://127.0.0.1:8000/RPC2" INFO:rtpp_command_ul_handle:GLOBAL: new session 9a1060af660d4202976c419ab5300470, tag 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 requested, type strong INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: new session on a port 35014 created, tag 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq;1 ERR:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: invalid socket name 1ea06d1eb0a8443697a08e568045598c;1 INFO:rtpp_command_ul_handle:9a1060af660d4202976c419ab5300470: pre-filling caller's address with 58.65.176.42:4022 DBUG:rtpc_doreply:GLOBAL: sending reply "35014 51.38.201.3 " DBUG:get_command:GLOBAL: received command "19086_4 D 9a1060af660d4202976c419ab5300470 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq 1ea06d1eb0a8443697a08e568045598c" INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully deleting session 1 on ports 35012/0 INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 35012/0 is cleaned up INFO:handle_delete:9a1060af660d4202976c419ab5300470: forcefully deleting session 1 on ports 35014/0 INFO:remove_session:9a1060af660d4202976c419ab5300470: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session:9a1060af660d4202976c419ab5300470: session on ports 35014/0 is cleaned up DBUG:rtpc_doreply:GLOBAL: sending reply "0 " DBUG:get_command:GLOBAL: received command "19086_5 D 9a1060af660d4202976c419ab5300470 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq 1ea06d1eb0a8443697a08e568045598c" INFO:handle_command:GLOBAL: delete request failed: session 9a1060af660d4202976c419ab5300470, tags 9RPsEBbcZSA23MmWAw0hfXzJ-bG1GYDq/1ea06d1eb0a8443697a08e568045598c not found DBUG:rtpc_doreply:GLOBAL: sending reply "E50 But if I comment out the param "Timeout_Socket" then there is no issue in bridging the media. Regards Hamid R. Hashmi

3 6

uac usage
by David Villasmil 21 Sep '18

21 Sep '18

Hello guys On my working config, i enabled uac like so: loadmodule "uac.so" ... modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_contact_addr", "test-proxy.singlecomm.com") ... And i added a user to the 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 1,10001,myuser,test-proxy.test.com,myuser,myprovider.com,myprovider.com ,myuser,mypass,,sip:myprovider.com,300,0,0 But it's not trying to register, and when i do: #> kamcmd uac.reg_info l_uuid 10001 I get: error: 404 - Record not found What am I doing wrong? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Jsonrpc_exec does not get body of request
by Denys Pozniak 21 Sep '18

21 Sep '18

Hello! I need to get some data via RPC to Kamailio inside by external event. But it seems long values of body brakes something: loadmodule "xhttp.so" loadmodule "jsonrpcs.so" modparam("jsonrpcs", "transport", 0) 1) event_route[xhttp:request] { ... jsonrpc_exec('{"jsonrpc": "2.0", "method": "core.psx", "id": 1}'); xlog("L_ERR","RPC code: $jsonrpl(code) \n"); xlog("L_ERR","RPC text: $jsonrpl(text) \n"); xlog("L_ERR","RPC body: $jsonrpl(body) \n"); return; } But in result I do not have body: Sep 21 14:27:55 /usr/sbin/kamailio[16140]: ERROR: <script>: RPC code: 200 Sep 21 14:27:55 /usr/sbin/kamailio[16140]: ERROR: <script>: RPC text: OK 2) If I request core.uptime I have something in body: event_route[xhttp:request] { ... jsonrpc_exec('{"jsonrpc": "2.0", "method": "core.uptime", "id": 1}'); xlog("L_ERR","RPC code: $jsonrpl(code) \n"); xlog("L_ERR","RPC text: $jsonrpl(text) \n"); xlog("L_ERR","RPC body: $jsonrpl(body) \n"); return; } Sep 21 14:37:17 /usr/sbin/kamailio[22683]: ERROR: <script>: RPC code: 200 Sep 21 14:37:17 /usr/sbin/kamailio[22683]: ERROR: <script>: RPC text: OK Sep 21 14:37:17 /usr/sbin/kamailio[22683]: ERROR: <script>: RPC body: {"jsonrpc":"2.0","result":{"now":"Fri Sep 21 14:37:17 2018","up_since":"Fri Sep 21 14:37:16 2018","uptime":1},"id":1} [root@kamailio]# kamailio -v version: kamailio 5.1.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. -- BR, Denys Pozniak

1 0

Include parameters in replies
by Duarte Rocha 21 Sep '18

21 Sep '18

Greetings, I need to to include some routing information (Two Id's, example "id_in=10;id_out=72") to be read later by Kamailio in every Request (Except the first one and in every Reply. For the requests, i'm including this info as parameters in the Record-Route and i'm reading it later in the Route Headers. For replies i thought about using the Via header. I'm using subst_hf() and i'm having success adding the parameter to Via in the requests. However, it does not seem the best idea since i breake the routing when i add the parameter. I wanted to avoid using custom headers for this since i can't really be sure that they won't be erased by any client's CPE or endpoint. Any ideas on this ? Thanks Cheers, Duarte Rocha

1 0

usr location
by volga629＠networklab.ca 21 Sep '18

21 Sep '18

Hello Everyone, User location lookup looks like can't handle long $rU like 10102-ce72256df4945bc472ed9c27a1037f46. It always return -3 404 not found. here relevant log Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: [INVITE] Call going from LAN SourceIP <10.18.130.50> to WAN. Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Looking user --> 10102-ce72256df4945bc472ed9c27a1037f46 in db Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Got [INVITE] and status <null> route[LOCATION] { $avp(oexten) = $rU; xlog("L_INFO", "Looking user --> $rU in db\n"); if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: route(MSG_STORE); # Meaning SMS has being stored for offline delivery if(!t_check_status("202")) { xlog("L_INFO", "Got [$rm] and status [$rs]\n"); # For rest traffic proceed as normal send_reply("404", "Not Found"); } exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } } kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 volga629

4 3

Re: [SR-Users] Daemon won't start
by Henning Westerholt 20 Sep '18

20 Sep '18

Am Samstag, 15. September 2018, 20:00:39 CEST schrieb hexhaxtron: > I can't make it work... > > It seems to have created the databases without any problem. > [pastebin cfg and logs] Hi Hexatron, please keep the list in CC. you should investigate why this error happens: Sep 14 06:52:07 thunix.org mysqld[1943]: 2018-09-14 6:52:07 140187532334848 [Warning] Access denied for user 'kamailio'@'localhost' (using password: YES) Sep 14 06:52:07 thunix.org /usr/sbin/kamailio[4254]: ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) Can you login with a mysql client on a shell to the database with the same credentials? Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

1 0

Kamailio On Tap – Social Event In Dusseldorf, Sep 27, 2018
by Daniel-Constantin Mierla 20 Sep '18

20 Sep '18

Hello, we are organizing an open social networking event in the evening of Sep 27, 2018, in Dusseldorf, at the end of the first day of Kamailio Developers meeting. Anyone in the area that wants to enjoy beer, pizza and chat about Kamailio, VoIP/RTC is welcome to join us. More details at: * https://www.kamailio.org/w/2018/09/kamailio-on-tap-social-event-in-dusseldo… Note that participation is free, but you have to announce your participation until Sep 26 in order to plan properly the event. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 0

usr location
by volga629＠networklab.ca 20 Sep '18

20 Sep '18

Hello Everyone, User location lookup looks like can't handle long $rU like 10102-ce72256df4945bc472ed9c27a1037f46. It always return -3 404 not found. here relevant log Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: [INVITE] Call going from LAN SourceIP <10.18.130.50> to WAN. Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Looking user --> 10102-ce72256df4945bc472ed9c27a1037f46 in db Sep 19 23:37:26 canlvprx01 /usr/sbin/kamailio[29217]: INFO: <script>: Got [INVITE] and status <null> route[LOCATION] { $avp(oexten) = $rU; xlog("L_INFO", "Looking user --> $rU in db\n"); if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: route(MSG_STORE); # Meaning SMS has being stored for offline delivery if(!t_check_status("202")) { xlog("L_INFO", "Got [$rm] and status [$rs]\n"); # For rest traffic proceed as normal send_reply("404", "Not Found"); } exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } } kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 volga629

1 0

re- double record route
by Wilkins, Steve 19 Sep '18

19 Sep '18

Hi Henning, Yes I do have that enabled. What is happening is that one of the providers on IOS is sending a double record route on the INVITE, but it is getting lost somewhere so when I send a 'BYE', I get a "404 not here". When I look at the SIP message I see only one of the record routes from the INVITE, therefore my assumption is, this is what is causing the 404, because everything else looks good.

3 8

occasional processing delay and UAC remote registration
by Dmitri Savolainen 19 Sep '18

19 Sep '18

Hi all. i encountered in my 4.1.5 kamailio server some inconstant issue. I have 2 listen address: addr1 for incoming SIP requests and as source addr for remote registration (via UAC module) addr2 for incoming SIP requests only uac config for regs: modparam("uac", "reg_keep_callid", 1) modparam("uac", "reg_random_delay", 120) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_timer_interval", 40) i have response time checking tool, that send OPTIONS every second, which immedeately replied by sl_send_reply("200") without any processing. Usualy respose time <1ms. Sometimes (2-3 per day) addr1 stop to response for these OPTIONS (timeout >5sec), while addr2 continue responding with time <1ms. In kamailio.log at this moment bunch of messages like ERROR: uac [uac_reg.c:1034]: uac_reg_tm_callback(): got sip response 403 > while registering ERROR: uac [uac_reg.c:1034]: uac_reg_tm_callback(): got sip response 408 > while registering That is normal: some remotes are broken. The problem is: 1. usually these messages appearing is not the reason of problem, but all problems are while these messages in log 2. it may be on OFF peak / ON peak day period 3. in my test installation i have much more remote failure registration (X 10), and the problem is not manifested more often (but i can catch it 2-3 per day) I think adding more workers for addr1 may help (now 8 worker and 16 CPUs). But it is interesting to understand the source of this issue -- Savolainen Dmitri

1 2

Where does location ruid come from?
by David Cunningham 19 Sep '18

19 Sep '18

Hello, We are debugging an issue with location records having duplicate "ruid" values, which aren't allowed by the table index. Can anyone tell us where the "ruid" value comes from? Is it a function of some other variable? Thanks in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

Can't get from cache: $sqlrows(asg)
by David Villasmil 18 Sep '18

18 Sep '18

Hello guys, i've got this in my config: Module config: modparam("sqlops","sqlcon","asg=>mysql://DB_USER:DB_PASS@DB_HOST/kamailio") Route: sql_query("asg", "insert ignore dispatcher (id, setid, destination, flags, priority, attrs, description) values (NULL, $var(dispatcher_id), 'sip:$si:5080', 0, 0, 'weight=$var(new_weight);maxload=50', '$fU')"); xlog("L_ERR", "[ASG_ADD_TO_DISPATCHER] Affected rows: $sqlrows(asg)\n"); if ($sqlrows(asg)>0) { ds_reload(); } else { xlog("L_ERR", "[ASG_ADD_TO_DISPATCHER] COULD NOT ADD THE NEW GATEWAY WITH IP $si AND USERNAME $fU\n"); } and i'm getting: 0(3929) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1261, column 20-32: Can't get from cache: $sqlrows(asg) I can't figure out where the problem might be. Can anyone help? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 4

Await to INVITE after REGISTER for mobile device which get awake via android/iOS notification messages
by Kristijan Vrban 18 Sep '18

18 Sep '18

Hi, now that many SIP endpoints are mobile devices which usually deep-sleep, and only get awake after Android/iOS own notification messages awake them, what are the best practices to await a a pending INVITE to such a mobile device until it REGISTER after the awake? I currently use a Asterisk a do a simple loop, trying new INVITES every second, but this is not elegant. A nice solution would be a type of queue, where the pending INVITE get send to the new location triggered immediately by the fresh REGISTER from that device to avoid any delay during call setup Kristijan

2 1

Incoming call on Kamailio WebRTC
by Adesh Pandey 17 Sep '18

17 Sep '18

Hi All, We are trying to setup WebRTC with Kamailio as sip registrar and dispatcher. We have succeed for outgoing calls but cannot attend incoming calls to the Kamailio. It would be a great help if someone can provide a sample config for webrtc + kamailio or blog for the same. Thanking you all -- Adesh Pandey Sr. Software Developer [image: phone] +91 92129 92129 Ext: 56 [image: moible] +91 8527384897 [image: Profile Pic] [image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter] <https://twitter.com/MyOperator>[image: Twitter] <https://www.youtube.com/user/MyOperatorCo>[image: Linkedin] <https://www.linkedin.com/company/voicetree-technologies>[image: Google+] <https://plus.google.com/+MyoperatorCo/posts> [image: Logo] <http://myoperator.co/>

2 1

Re: [SR-Users] Record-Route Parameters
by Duarte Rocha 17 Sep '18

17 Sep '18

Hey, I need all of my kamailio's IPs in Record Routes and i can do that already. What i want to do is add different parameters to each Record-Route. Is that possible? A sexta, 14/09/2018, 17:54, Duarte Rocha <duarterocha91(a)gmail.com> escreveu: > Greetings, > > My Kamailio has 2 IP's and, as such, it will generate two Record-Routes > when the Request is relayed. > > I want to add one parameter to one of the Record Routes and another > parameter to the other Record-Route. However, if i use add_rr_param() it > will automatically add the parameters to the two Record-Route Headers. > > Is there a way to do what i want without me re-writing the headers > directly ? > > Cheers, > > Duarte Rocha >

3 8

how to catch attacker using bad request line?
by Juha Heinanen 17 Sep '18

17 Sep '18

In order to be able to fail2ban an attacker that sends tons of SIP requests with malformed request lines, it would need to be possible to generate an appropriate syslog message from config file. I didn't find any sanity module param value that would turn on checking of request line syntax. Any other ideas? As an example, below is what comes to syslog when I send a request that has syntax error on request line. -- Juha Sep 17 14:46:39 char /usr/bin/sip-proxy[9458]: ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [(null)] Sep 17 14:46:39 char /usr/bin/sip-proxy[9458]: WARNING: <core> [core/receive.c:230]: receive_msg(): parsing relevant headers failed Sep 17 14:46:43 char /usr/bin/sip-proxy[9458]: ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [(null)] Sep 17 14:46:43 char /usr/bin/sip-proxy[9458]: WARNING: <core> [core/receive.c:230]: receive_msg(): parsing relevant headers failed

2 11

issue with rewrite to domain in failure route
by Yuriy Gorlichenko 17 Sep '18

17 Sep '18

Hi I'm building some system with failover trunk but my providers requires to set their doamin in domain part of To Header so when I rewrite it at the my failure_route $td = FAILOVER_PROVIDER my To filed looks like To: <sip: 123(a)my.failover.providermy.main.provider> what i tried: Move this change to BRANCH_ROUTE: here tried as $td = as uac_replace_to('',"newToFieldValue) Move to the my custom route with both variants But result is the same kamailio 5.1 version: kamailio 5.1.5 (x86_64/linux) d06080 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d06080 compiled on 00:43:08 Aug 23 2018 with gcc 6.3.0

5 11

Dispatcher Active IP for specific SetID over xHTTP
by Soltanici Ilie 17 Sep '18

17 Sep '18

How could I get the active IP from the dispatcher module for a specific setid by using xHTTP request?

1 0

double record route from INVITE
by Wilkins, Steve 16 Sep '18

16 Sep '18

Hi Mojtaba, But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE' (from IOS) need to be there, so that IOS can find it's Proxies and complete the transaction and send back a '200 OK'? Thank you,

1 0

double record route?
by Wilkins, Steve 16 Sep '18

16 Sep '18

Good Morning All, Is there any way to add a double record route? I tried adding a second record route and I always only get the first one added. I have tried record_route(....), and record_route_advertised_address(...), but I still only get the first record route added. Thank you, -Steve

2 1

Allow registration only for one client per AOR
by eyas barhouk 15 Sep '18

15 Sep '18

Hello dears, I'm using kamailio devel version as an IMS (P-CSCF,S-CSCC,I-CSCF) with forking mode ,when more than one client registered with the same AOR and the servers received a call to that Aor All the clients ringing , after that one of the clients answer the call which will cause cancel message with reason call completed elsewhere . This cancel message affect on the running call which answered by one of the client and terminate it . So how to set up kamailio to allow registration only for one client per Aor or to prevent the affection of this cancel messages . Thanks in advance & best regards

4 5

kamailio installable on stretch?
by David Villasmil 15 Sep '18

15 Sep '18

Hello guys, is kamailio from deb http://deb.kamailio.org/kamailio51 stretch main deb-src http://deb.kamailio.org/kamailio51 stretch main Installable on debain stretch? I'm getting: # apt-get install kamailio kamailio-xmlrpc-modules kamailio-mysql-modules Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'kamailio-xml-modules' instead of 'kamailio-xmlrpc-modules' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio : Depends: libreadline7 (>= 6.0) but it is not installable kamailio-mysql-modules : Depends: libmariadbclient18 (>= 5.5.36) but it is not installable E: Unable to correct problems, you have held broken packages. # but libreadline7 is not avaiable on stretch. Thanks David Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 3

Daemon won't start
by hexhaxtron 15 Sep '18

15 Sep '18

Hi! I did this tutorial: https://sonnguyen.ws/install-kamailio-debianubuntu/ When I try to start the daemon I get: # systemctl start kamailio Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. Doing systemctl status kamailio shows: # systemctl status kamailio ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Fri 2018-09-14 20:26:25 BST; 44s ago Process: 29482 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) Main PID: 24331 (code=exited, status=0/SUCCESS) Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: usrloc [dlist.c:654]: register_udomain(): failed to open database connection Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: registrar [reg_mod.c:566]: domain_fixup(): failed to register domain Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: <core> [route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:646 Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: ERROR: <core> [route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:649 Sep 14 20:26:25 thunix.org /usr/sbin/kamailio[29484]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Sep 14 20:26:25 thunix.org systemd[1]: kamailio.service: Control process exited, code=exited status=255 Sep 14 20:26:25 thunix.org systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Sep 14 20:26:25 thunix.org systemd[1]: kamailio.service: Unit entered failed state. Sep 14 20:26:25 thunix.org systemd[1]: kamailio.service: Failed with result 'exit-code'. My /etc/kamailio/kamailio.cfg is: http://paste.debian.net/1042245/ This is the output of journalctl --no-pager|grep kamailio http://paste.debian.net/1042246/ What should I do to fix this? Thanks, hexhaxtron

2 1

Record-Route Parameters
by Duarte Rocha 15 Sep '18

15 Sep '18

Greetings, My Kamailio has 2 IP's and, as such, it will generate two Record-Routes when the Request is relayed. I want to add one parameter to one of the Record Routes and another parameter to the other Record-Route. However, if i use add_rr_param() it will automatically add the parameters to the two Record-Route Headers. Is there a way to do what i want without me re-writing the headers directly ? Cheers, Duarte Rocha

2 1

Record Route Parameters
by Duarte Rocha 15 Sep '18

15 Sep '18

Greetings, I have Kamailio as an SBC with mutiple IP's and I need to save some information in the first Invite about the routing made internally in the Kamailio. I'm thinking about adding some parameters to Record-Route and retrieve them in the Route header. Is this safe ? Can I be sure that those parameters won't be deleted or changed? Best Regards, Duarte Rocha

3 2

maxbuffer value
by Dmitri Savolainen 14 Sep '18

14 Sep '18

Hi. Does anybody know what kamailio core maxbuffer parameter means? And is there any conjunction with system net.core.rmem_max parameter? Doc is obscure for me "The size in bytes not to be exceeded during the auto-probing procedure of descovering the maximum buffer size for receiving UDP messages. " -- Savolainen Dmitri

2 1

kamailio installable on stretch?
by David Villasmil Govea 14 Sep '18

14 Sep '18

Hello guys, is kamailio from deb http://deb.kamailio.org/kamailio51 stretch main deb-src http://deb.kamailio.org/kamailio51 stretch main Installable on debain stretch? I'm getting: # apt-get install kamailio kamailio-xmlrpc-modules kamailio-mysql-modules Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'kamailio-xml-modules' instead of 'kamailio-xmlrpc-modules' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio : Depends: libreadline7 (>= 6.0) but it is not installable kamailio-mysql-modules : Depends: libmariadbclient18 (>= 5.5.36) but it is not installable E: Unable to correct problems, you have held broken packages. # but libreadline7 is not avaiable on stretch. Thanks David -- DVG -- Imagination is more important than knowledge Albert Einstein ᐧ

1 0

Forward SIP request between two Kamailio servers
by Nathanael Eneroth 14 Sep '18

14 Sep '18

Hi,? I have a simulation setup of 4 VMs in Virtual Box - two UAs and two Kamailio servers. The first UA will register to KAM1, and the second UA will regsiter to KAM2. See the Figure below. UA1<--->KAM1<---->KAM2<---->UA2 int1 int1 int3 int3 int2 int2 I would like KAM1 to forward all SIP requests destined to int3 via KAM2 and vice versa. I am aware of the routing logic in 'kamailio.cfg', but i find rather excessive. Pseudo code of what i'm trying to achieve: if (is_method("INVITE")) { if(dst_ip == int3) set_next_kamailio_server(KAM2) } Or: if (is_method("INVITE")) { if(callee is unkown) ask_kamailio_server(KAM2) } Thanks for all the help

4 3

DMQ - Failed to get domain
by Daniel W. Graham 14 Sep '18

14 Sep '18

Receiving the following message on the node receiving usr_loc via DMQ, for each location received: DEBUG: dmq [worker.c:87]: worker_loop(): dmq_worker [0 54958] lock acquired DEBUG: dmq_usrloc [usrloc_sync.c:460]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@x.x.x.x DEBUG: dmq_usrloc [usrloc_sync.c:405]: usrloc_dmq_execute_action(): Received DMQ_UPDATE. Update contact info... ERROR: dmq_usrloc [usrloc_sync.c:70]: add_contact(): Failed to get domain Any input is appreciated. Thanks, -dan

2 10

Restart kamailio on critical error
by Sergey Safarov 14 Sep '18

14 Sep '18

Is possible to restart kamailio when is critical errors is happens? Now in log ``` Sep 13 18:50:38 ca-edm-sw3.phonebox.work sh[1171]: 81(85) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/85 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:743]: handle_sigs(): child process 5 exited by a signal 9 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:743]: handle_sigs(): child process 80 exited by a signal 9 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:743]: handle_sigs(): child process 110 exited by a signal 9 Sep 13 18:52:21 ca-edm-sw3.phonebox.work sh[1171]: 0(1) ALERT: <core> [main.c:746]: handle_sigs(): core was not generated ``` Some of childs is died but main kamailio process is still active. SIP traffic is not processed because TCP socket is died. When i core is generated i can force restart process. But i cannot parse logs to catch such event. May exist other way? Sergey

2 2

Python KEMI performance
by Aleksandar Sosic 14 Sep '18

14 Sep '18

Hi all, is there anyone here that has done a performance testing measurements between native kamailio config file and full python kemi instance of kamailio? I've read somewhere that the python kemi code was 30% slower. Why is that and are there any plans on trying to improve the performance of python kemi? Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

3 2

limit call per duration
by António Silva 13 Sep '18

13 Sep '18

Hi, I'm new to kamailio, i want to be able to limit call per duration, how can i do it? i found module call_control, http://www.kamailio.org/docs/modules/5.1.x/modules/call_control.html, do you recommend it to use? anyone using it that would like to share the experience, like the values for cpu usage, is it realtime, support for multiple calls per account, other... Thanks!! -- Saludos / Regards / Cumprimentos António Silva

1 2

How to use Geoip Module in Kemi
by Naveed Ur Rehman 13 Sep '18

13 Sep '18

Hi, I am trying to use geoip2 module in kemi using LUA, The issue is that when i try to access the $gip2(src=>cc) variable using lua, the kamailio instance killed and stops, this is my code snippet, kindly advise anyone how this can be sorted local sourceIP=KSR.pv.get("$si"); if KSR.geoip2.match("sourceIP","src") then local x= KSR.pv.get("$gip2(src=>cc)"); end thanks Engineer Naveed Ur Rehman 0923415652825 Computer Systems Engineering

2 1

TOPOH module im REGISTER method.
by Mojtaba 13 Sep '18

13 Sep '18

Hello, I am used topoh module in kamailio, Why this module does not work (No change ) in Register request Method? I need to change UA' ip address in Via header in Register request before forwarded to my network? Does the topoh module used in this situation? I have not this issue in INVITE request, I just have this issue in Register request. Thanks. With Regards.Mojtaba -- --Mojtaba Esfandiari.S

3 5

Crashed drouting
by Евгений Голей 12 Sep '18

12 Sep '18

Good evening. I have errors on the server (kamailio v 5.1.0) "kamailio [30529]: segfault at 0 ip 00007f9c00c29cfb sp 00007fffa9808310 error 4 in drouting.so [7f9c00c26000 + 39000]" which cause the service to stop. Although the drouting module has been used for a long time, this has never happened. This was done after adding the functions of the dispatcher module. What could be the reason ? Example error in system kamailio [20728]: segfault at 0 ip 00007f32c1c8fcfb sp 00007fff78430d00 error 4 in drouting.so [7f32c1c8c000 + 39000] in the logs of kamailio 018-09-08T22: 30: 29.0448 "ALERT: <core> [main.c: 746]: handle_sigs (): child process 10556 exited by a signal 11" 2018-09-08T22: 30: 29.0452 "ALERT: <core> [main.c: 749]: handle_sigs (): core was generated" 2018-09-08T22: 30: 29.0456 "INFO: <core> [main.c: 771]: handle_sigs (): terminating due to SIGCHLD" -- B.R. Evgeniy

3 7

remove_hf("Bla") for all requests
by Sergiu Pojoga 12 Sep '18

12 Sep '18

Hi there, Say I need to remove_hf("User-Agent") for all requests, back and forth. So I add it at the top of *request_route *section. However, replies don't seem to be affected by it. Do I really need to arm a *t_on_reply route* for this simple purpose? "Problem" with that is that *route[RELAY]* already has some *onreply_route* block doing things like *NATMANAGE *for some methods and as far as I know - only one* onreply_route* can be armed for a request? route[RELAY] { ... if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } t_relay(); } onreply_route[MANAGE_REPLY] { ... route(NATMANAGE); } Any suggestions? Thanks.

4 7

Re: [SR-Users] kamailio + rtpengine + custom filename
by Жан Базаров 12 Sep '18

12 Sep '18

Message-ID: <CABVi_EzZRF8nUqoBGcUtUeAV2pknn5S0urcmJFgHSSz6N1kJ5g(a)mail.gmail.com> Content-Type: text/plain; charset="UTF-8" Hello, You could use metadata to save your favourite filename on it. After recording is finished, replace it. The metadata file is created based on <CALL_ID> for each call. With Regards. Mojtaba On Mon, Sep 10, 2018 at 8:53 PM Жан Базаров <chiefkeeft(a)gmail.com> wrote: > > Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> > How i can change filename in start_recording() ??? Thx you. Can you help me with it task? Show me, please, an example. I want filename of the following format, for example: 'output-dir/yyyy/mm/dd/fromnumber/callid-tonumber-uniqueid.wav'.

2 1

Missing ACC record in case of canceled call forking
by Marco Capetta 12 Sep '18

12 Sep '18

Hi All, I'm facing a strange problem of missing ACC record in case of parallel call forking. The scenario is the following: - A subscriber with 1 device registered - B subscriber with 2 device registered (B1 and B2) CASE 1: - A calls B - B1 and B2 start ringing - A hangups the call before B1 or B2 answers Kamailio generates an ACC record. CASE 2: - A calls B - B1 and B2 start ringing - B1 rejects the call sending back a 486 - B2 is still ringing - A hangups the call before B2 answers Kamailio DOESN'T generate an ACC record. We have Kamailio v5.1.4 with TM module enabled. ACC configuration is the following: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) I increased debug level of TM and ACC modules and I added some debug lines as well. The difference between the two calls is that, after the CANCEL sent by A is processed by Kamailio, in CASE 1 I have the following lines: Aug 28 14:21:11 spce proxy[13188]: DEBUG: tm [t_hooks.c:258]: run_trans_callbacks_internal(): DBG: trans=0x7f5a88049198, callback type 512, id 0 entered Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_logic.c:670]: tmcb_func(): acc callback called for t(0x7f5a88049198) event type 512, reply code 487 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 0 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 1 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 2 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 3 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 4 Those debug lines are not printed for CASE 2. Is there any configuration I'm missing or is it a bug? Thank you very much for your support. Marco

2 14

Call termination via dialog RPC dlg.end_dlg for the SIP-UA behind NAT
by Denys Pozniak 11 Sep '18

11 Sep '18

Hello! When calling RPC command dlg.end_dlg, BYE for leg behind NAT sends according to the Contact header not to the real ip:port. RPC ul.dump shows correct address in Received: sip:2.2.2.2:7488, but looks dialog does not know real address. How can I force sending to the real address? [root@proxy1]# kamcmd dlg.list { h_entry: 3753 h_id: 6741 call-id: 0_618427980(a)10.10.10.10 from_uri: sip:1234567890@1.1.1.1:5060 to_uri: sip:3333333333@1.1.1.1:5060 state: 2 start_ts: 0 init_ts: 1536671492 timeout: 0 lifetime: 7200 dflags: 512 sflags: 4096 iflags: 32 caller: { tag: 2750096738 contact: sip:1234567890@10.10.10.10:5060 cseq: 2 route_set: socket: udp:1.1.1.1:5060 } callee: { tag: contact: cseq: route_set: socket: } profiles: { } variables: { } } [root@proxy1]# kamcmd ul.dump Info: { AoR: 1234567890 HashID: 262653578 Contacts: { Contact: { Address: sip:1234567890@10.10.10.10:5060 Expires: 1269 Q: -1.000000 Call-ID: 0_213504837(a)10.10.10.10 CSeq: 8 User-Agent: Yealink SIP-T21P Received: sip: 2.2.2.2:7488 Path: [not set] State: CS_NEW Flags: 0 CFlags: 65600 Socket: udp: 1.1.1.1:5060 Methods: 16383 Ruid: uloc-5b97c358-6052-1 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1536674159 Last-Modified: 1536674159 } } } -- BR, Denys Pozniak

1 0

Remove branch from branch list
by Yuriy Gorlichenko 11 Sep '18

11 Sep '18

Hi I'm trying to manipulate with branches at the $branch variable. 8 actually need to remove unneeded branch if not passed some checks. For now I Just rewriting uri and dst_uri with some face address but it looks little bit dirty way. Is here any an other more clear way to do this?

2 2

Record Routes
by Wilkins, Steve 11 Sep '18

11 Sep '18

Hello All, Can a Route: be removed?, if so, how? Thank you?

2 4

Logging of messages generated by Kamailio itself
by Konstantin Polyakov 11 Sep '18

11 Sep '18

Hello, Could somebody help with one issue? In some cases Kamailio can send messages itself. For example, when fr_inv_timer is expired Kamailio sends 408 to the INVITE sender and CANCEL to the INVITE receiver. These two messages (408 and CANCEL) are not printed into the log and I cannot find a way to do that. There are also other scenarios when some Kamailio modules send SIP messages by its own. So is there way to print out into log all such messages? My goal is to get full message flow in the log per certain call-id. Thank you in advance. Best regards. Konstantin

2 9

kamailio + rtpengine + custom filename
by Жан Базаров 11 Sep '18

11 Sep '18

Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> How i can change filename in start_recording() ???

2 1

tpops module
by volga629＠networklab.ca 11 Sep '18

11 Sep '18

Hello Everyone, I am using tcpops module for keepalive and on BYE when tcp keeplaive close is called produce this error in log. Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: <core> [core/sr_module.c:1781]: get_int_fparam(): Could not convert PV to int Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: tcpops [tcpops_mod.c:293]: w_tcp_keepalive_disable1(): invalid parameter 'con' (must be a number) kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 if (is_method("BYE|CANCEL")) { $avp(bye_conid) = $conid; rtpengine_delete(); t_on_reply("4"); } onreply_route[4] { if(is_method("BYE") && t_check_status("200")) { tcp_keepalive_disable("$avp(bye_conid)"); } } volga629

3 3

Simultaneous usage of t_on_reply() and t_on_failure()
by Володимир Іванець 10 Sep '18

10 Sep '18

Hello everyone, I wanted to ask those who know if Kamailio's behavior I'm facing is expected or I should make some improvements to the configuration. Kamailio version is 5.1.0. I have a route where RTPEngine parameters are being collected and *rtpengine_offer()* is called. After that *t_on_reply("REPLY_SIP_TO_SIP");* followed by the *t_on_failure("FAILURE_SIP_TO_SIP");* are used. The idea is to process all responces except 415 or 488 from UAC as usual in *onreply_route[REPLY_SIP_TO_SIP]* and use *failure_route[FAILURE_SIP_TO_SIP]* to update SDP with *rtpengine_offer()* if necessary. *onreply_route[REPLY_SIP_TO_SIP]* just goes to *exit;* if *$rs* equals 415 or 488. This works fine with Htek phone which sends 100, 180 and then 488. But I can not see *failure_route[FAILURE_SIP_TO_SIP]* execution for calls to Zoiper which replies with 100 and immediately 415. t_on_failure(failure_route) documentation says: "Sets failure routing block, to which control is passed after a transaction completed with a negative result but before sending a final reply." and to be honest I don't really understand how lacking of responce prevents failure_route from executing. Thanks a lot!

4 5

Crashed kamailio drouting
by Евгений Голей 08 Sep '18

08 Sep '18

Good evening. I have errors on the server (kamailio v 5.1.0) "kamailio [30529]: segfault at 0 ip 00007f9c00c29cfb sp 00007fffa9808310 error 4 in drouting.so [7f9c00c26000 + 39000]" which cause the service to stop. Although the drouting module has been used for a long time, this has never happened. This was done after adding the functions of the dispatcher module. What could be the reason ? Example error in system kamailio [20728]: segfault at 0 ip 00007f32c1c8fcfb sp 00007fff78430d00 error 4 in drouting.so [7f32c1c8c000 + 39000] in the logs of kamailio 018-09-08T22: 30: 29.0448 "ALERT: <core> [main.c: 746]: handle_sigs (): child process 10556 exited by a signal 11" 2018-09-08T22: 30: 29.0452 "ALERT: <core> [main.c: 749]: handle_sigs (): core was generated" 2018-09-08T22: 30: 29.0456 "INFO: <core> [main.c: 771]: handle_sigs (): terminating due to SIGCHLD" -- BR Evgeniy

1 0

kamailio.cfg errors with this example from documentation
by Wilkins, Steve 07 Sep '18

07 Sep '18

$var(x) = "abcd"; $(var(x){s.substr,1,0}); Hello all, I took the example from the documentation, but Kamailio.cfg has errors with this example. Errors=> 0(14444) CRITICAL: <core> [core/cfg.y:3489]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 1-23: pvar with transformations in assignment left side 0(14444) CRITICAL: <core> [core/cfg.y:3492]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: syntax error 0(14444) CRITICAL: <core> [core/cfg.y:3492]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: bad command Thank you,

2 5

Via header line missing
by Karlheinz Knapp 07 Sep '18

07 Sep '18

Hi, hm.... the only reason why I'm playing with kamailio is that it should be able to translate IPv6 from customer phones behind Fritz!Box (NAT) to IPv4 used by the provider (which is not able to handle IPv6). But it looks that he accept in general the packet with 2 Via lines (also with IPv6 as address) and send a normal reply. The second Via is not needed, because he have to send to the first Via. And if it really comes to trouble, I can remove it with kamailio :) If it is not possible to make a complete translation with kamailio, then I have to search again for an other solution. But still: why is this one Via line not transmitted on the IPv6 side? Only when I do some modifications it appears magically. That's not a normal behaviour in my opinion. If you or Henning can not reproduce this behaviour, then it's possible that I start TeamViewer and I can show this on our installation (only at normal working hours). Btw. I use sngrep to display all the messages, but it works stable and reliable. Best regards, i.A. Bernd

1 0

Event: Kamailio Developers Meeting, Sep 27-28, 2018, in Dusseldorf, Germany
by Daniel-Constantin Mierla 07 Sep '18

07 Sep '18

Kamailio SIP Server project is organizing a meeting of its developers during September 27-28, 2018, hosted by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join. Please note we have a limited capacity of 20 seats in the meeting room. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: - https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 1

kamailio with evapi crashing on high volume
by Jayesh Nambiar 07 Sep '18

07 Sep '18

Hello, I'm using kamailio 5.1.5 with evapi. I have a node.js connecting with kamailio evapi to which I send events and also consume events based on which I do the routing. I have 8 evapi workers defined in the config. The problem is that kamailio randomly crashes on high load. I'm assuming that it is related to the evapi module as rest of the config is pretty straight forward. I could get a core file and here's the core dump: [New LWP 14042] [Thread debugging using libthread_db enabled] Using host libthread_db library “/lib/x86_64-linux-gnu/libthread_db.so.1”. Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/siptrunk.pid -f /usr/local/carrie’. Program terminated with signal SIGABRT, Aborted. #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54 54 ../sysdeps/unix/sysv/linux/raise.c: No such file or directory. (gdb) bt #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54 #1 0x00007f999528502a in __GI_abort () at abort.c:89 #2 0x000000000041a029 in sig_alarm_abort (signo=14) at main.c:646 #3 <signal handler called> #4 0x00007f999534f497 in __libc_cleanup_routine (f=<optimized out>) at ../sysdeps/nptl/libc-lockP.h:291 #5 closelog () at ../misc/syslog.c:415 #6 0x0000000000000000 in ?? () Any help in this regards is would allow me to identify the reason of the crash. Thanks for the support. - Jayesh

2 5

Kamailio with CGNAT
by Amar Tinawi 06 Sep '18

06 Sep '18

Hello Although my question it's not related to Kamailio directly, but i think someone could help in this strange behavior in our tests, we have some clients under private networks, where the provider of those networks implements CGNAT (NAT444) for it's customers to reach internet (Client Private IP) --->(Home router: Private to Private NAT)--->(FW with CGNAT)---------------------------->(Kamailio Proxy Public IP) Registration is performing fine, but making calls from this client had strange problem as follow: Client -------------INVITE--------------> Proxy Proxy -------------Trying---------------->Client Proxy--------------INVITE---------> Called Party The called party should send Ringing msg, but the massage get lost somewhere and didn't reach to the Proxy. Although all the IP's in sip massages are public i tried with STUN/TURN and NAThelper module solutions, but didn't help is there any way to slove CGNAT with SIP in Kamailio? Thanks in Advance

4 11

dlg.end_dlg on RTP timeout
by Igor Olhovskiy 06 Sep '18

06 Sep '18

Hi all! Is there any way to end dialog in Kamailio on RTP Timeout event on RTPEngine? Or only look at logs/redis database with external tool? Regards, Igor

5 13

call_stats
by David Villasmil 06 Sep '18

06 Sep '18

Hello all, Anyone got the table definition for call_stats? I'm getting db_mysql_submit_query(): driver error on query: Table 'kamailio.call_stats' doesn't exist (1146) And i can't find the table anywhere. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 2

Via header line missing
by Karlheinz Knapp 06 Sep '18

06 Sep '18

Hi, of course kamailio is listening on IPv4 and IPv6. Register coming from IPv6 and goes out on IPv4: REGISTER sip:sip.ggew-net.de SIP/2.0 Via: SIP/2.0/UDP 217.113.176.155;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 Via: SIP/2.0/UDP [2A05:E2C2:0:401:F40D:E8C5:5D85:2A95]:5060;received=2A05:E2C2:0:401:F40D:E8C5:5D85:2A95;branch=z9hG4bK001755af58b0e811a048bb9f393b8bfe;r rt=5060 From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de> Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER Contact: <sip:4962519448281@ [2A05:E2C2:0:401:F40D:E8C5:5D85:2A95]:5060>;+sip.instance="<urn:uuid:00E0F96C-5BAF-E811-A877-86B14B4F5249>" Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK Max-Forwards: 69 Allow-Events: org.3gpp.nwinitdereg User-Agent: SIPPER for PhonerLite Supported: replaces, from-change, gruu Expires: 900 Content-Length: 0 P-hint: outbound Answer on IPv4: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 The same answer on th IPv6 side after kamailio without something special: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 It contains no Via header!!! If I add my append_hf() append_hf("Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=$sel(via[1].branch);rport=5060\r\n", "To"); msg_apply_changes(); I get: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 With the original Via header !!! If I add a second Via with append_hf() append_hf("Via: SIP/2.0/UDP [2a05:e2c2:0:401:3002:AA:754A:F475]:5060;rport\r\n", "To"); append_hf("Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=$sel(via[1].branch);rport=5060\r\n", "To"); msg_apply_changes(); I can see this additional line in the IPv6 packet too. SIP/2.0 401 Unauthorized Call-ID: 0038C332-5AB0-E811-ABC9-80F5AA6CB72C@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=2618219059 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07871-795b4386-0dc31d606 Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=z9hG4bK0ce8.b94e5d3389049004487a8551ac2c8c41.0;rport=5060 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK0ce8.b94e5d3389049004487a8551ac2c8c41.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="795b3345266c3dd57a52e3237f4276dd",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 But since I don't have the right branch, PhonerLite starts a new registration instead of reregister with username password. Best regards

2 1

null $au
by David Villasmil 06 Sep '18

06 Sep '18

Hello, in the block: 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(); } I'm trying to print out $au, but it's coming up NULL many times, i don't get why this is. If i remember correctly, the script goes through this block twice, once when we get the REGISTER with no credentials, and once when the REGISTER comes with credentials after the challenge... is this correct? i need to know how to catch $au when the authentication is successful... help? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 2

Via header line missing
by Karlheinz Knapp 05 Sep '18

05 Sep '18

Hi, we want to use kamailio as a IPv6 to IPv4 proxy. IPv4 to IPv4 bridging with NAT works. If we come from an IPv6 address, the registration packet is send via IPv4 to the provider. He answers with 401 Unauthorized and one Via line in the header. On the IPv6 side was no Via line in the packet. So I tried to add one line by hand using an onreply_route and append_hf(), msg_apply_changes() To my suprise, this line does never appear, but the original Via line is now in the packet !!! If I add now additional Via header lines, they appear in the packet too. This looks realy like a bug to me. A solution is needed, because the phone does not respond to the 401 with user and password. It starts a new registration instead. Maybe I can fix it with my additional Via linem but I need the original branch ID (I think) How can I get this? Btw. we have this behaviour with 4.4.2 and 5.1.5 If I get no answer during the next 12 hours I'll try 3.3.7 Best regards

2 1

dispatcher local socket
by Евгений Голей 05 Sep '18

05 Sep '18

Hello! Kamilio version 5.1.0 Please tell me why the module dispatcher does not correctly determine the availability of destination gateways. The scheme is as follows: Kamailio (udp:XX.XX.XX.XX:5060) ----- Provider_1 -----> (sip:aa.bb.cc.dd:5060) Dest GW Kamailio (udp:YY.YY.YY.YY:5060) ------- Provider_2 -----/ In table dispatcher: setid | destination | flags | priority | attrs 17 | sip:aa.bb.cc.dd:5060 | 8 | 50 | socket=udp:XX.XX.XX.XX:5060 17 | sip:aa.bb.cc.dd:5060 | 8 | 40 | socket=udp:YY.YY.YY.YY:5060 That is, our server controls the availability of the remote gateway through two local sockets, in order to switch to the backup if it is not available through the primary Internet provider. But as I understand at the moment when determining the availability of a remote gateway, the outgoing socket does not play any role, that is, if the remote gateway is not accessible through the secondary operator (priority is less than the primary one), only the one that has priority above is always inactive. Are there any solutions to this problem? BR. Evgeniy

1 0

From URI attribute for dispatcher target
by Giacomo Vacca 05 Sep '18

05 Sep '18

Hi all, I was looking for a solution to set a dedicated From URI in OPTIONS requests to a specific dispatcher target. ds_ping_from ( https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc…) applies globally, why I needed the granularity of a single target. With these changes to the dispatcher module I get what I need: https://github.com/kamailio/kamailio/commit/3f9c38be11416f7abe5c4448088ba66… I can just add an attribute to the others, e.g. instead of *1 sip:127.0.0.1:7070 <http://127.0.0.1:7070> 0 1* something like: *1 sip:127.0.0.1:7070 <http://127.0.0.1:7070> 0 1 ping_from=sip.custom.com <http://sip.custom.com>* When the ping_from attribute is not specified, the general ds_ping_from applies and is used. This would save me from a solution like intercepting the outgoing OPTIONS and manipulating the From header. Does this approach make sense? Are there other ways already available to achieve this? Thanks, Giacomo

1 0

SCA: sca_call_info_bye_handler: no in-use callee appearance for BYE
by Asgaroth 05 Sep '18

05 Sep '18

Hi All, I've just started playing with the SCA module and am encountering an issue whereby SCA cannot seem to find an in-use appearance for the BYE message. From my limited understanding of SCA, the module, in this case, appears to be trying to track the state of the call using callid as there are no call-info headers in the signalling for it to extract and use. I call sca_call_info_update() for initial invite, 100 response, 180 response, 200 response and again for the terminating bye. When I answer the call, at the 200 ok for the initial invite, I can see sca sending a notify to the subscribed devices indicating appearance state being active. I can verify this with kamcmd sca.all_appearances. However, when I hang up the call, when sca_call_info_update() is fired for the bye message it fails with the following error message: sca [sca_call_info.c:1587]: sca_call_info_bye_handler(): sca_call_info_bye_handler: no in-use callee appearance for BYE <aor> from <from uri>, call-ID 7989c223-2ba4-1237-1888-20474786734e I cannot seem to track down where this is failing for me, or where I may be missing something in my configuration. sca.all_appearances does appear to have the entry mentioned in the failure message, there must be something I'm missing somewhere. Is there anyone out there who is using the sca module and may have a moment to assist with this issue. I have a pcap trace that I can send privately. Any assistance would be greatly appreciated. Thanks

1 0

Kamailio v5.1.5 Released
by Daniel-Constantin Mierla 04 Sep '18

04 Sep '18

Hello, Kamailio SIP Server v5.1.5 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.5. For more details about version 5.1.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/08/kamailio-v5-1-5-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.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 1

Problem with reconnection to database server
by Minh Phan 04 Sep '18

04 Sep '18

Hi List, My Kamailio instance is disconnected from database server (Postgres) quite often (after running for some days) and it won't manage to reconnect. Any idea why this happens? Normally it try to reconnect when connection with the DB is lost, desn't it? Bellow are the related log messages. Best regards, Minh Phan Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR, Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:609]: db_postgres_insert(): unexpected result returned Sep 4 11:31:20 : ERROR: topos [tps_storage.c:599]: tps_db_insert_dialog(): failed to store message Sep 4 11:31:20 : ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK66fa.3f007792000000000000000000000000.0' Sep 4 11:31:20 : ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: insert into topos_d (rectime,a_callid,a_uuid,b_uuid,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag,a_rr,b_rr,s_rr,iflags,a_uri,b_uri,r_uri,a_srcaddr,b_srcaddr,s_method,s_cseq ) values ('2018-09-04 11:31:20','1ff8d1d7606f7416-26611(a)192.168.70.106 ','','','','','','','ae86e22a93159b5c6ac75234cdf26c1b-44e5','','','','',0,'','','','','','OPTIONS','10') Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 kamailio[26608]: ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK86fa.424ded00000000000000000000000000.0' Sep 4 11:31:20 kamailio[26608]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 kamailio[26608]: ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR, Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:609]: db_postgres_insert(): unexpected result returned Sep 4 11:31:20 : ERROR: topos [tps_storage.c:599]: tps_db_insert_dialog(): failed to store message Sep 4 11:31:20 : ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK76fa.d33cc801000000000000000000000000.0' Sep 4 11:31:20 : ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: insert into topos_d (rectime,a_callid,a_uuid,b_uuid,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag,a_rr,b_rr,s_rr,iflags,a_uri,b_uri,r_uri,a_srcaddr,b_srcaddr,s_method,s_cseq ) values ('2018-09-04 11:31:20','1ff8d1d7606f7417-26611(a)192.168.70.106 ','','','','','','','ae86e22a93159b5c6ac75234cdf26c1b-917f','','','','',0,'','','','','','OPTIONS','10') Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR,

2 1

DMQ mem leak issues
by Rogelio Perez 04 Sep '18

04 Sep '18

Hello, We're running three instances of Kamailo v5.14 as registrars handling registrations from ~2000 SIP clients, with one instance being primary and the other two as backups. The three of them are using the dmq and dmq_usrloc modules to synchronize user locations, however after a couple of days of operation the two failover instances show memory leak behaviors, with mem usage assigned to the core taking all available resources. When this happens we've noticed that: - The shared memory used by the function "sip_msg_shm_clone" spikes (from 1kb to 1.5GB). - The shared memory used by the function "dmq:worker.c:job_queue_push" also increases, but not as much (from 1kb to 1MB) - DMQ request are not being answered (with a 200 OK) by the affected instance during this memory leak, which make us think that DMQ module becomes unresponsive. A few more notes: - The failover instances are doing nothing except receiving replicated contacts. - The shared memory grows at the same rate on both instances, but the critical behavior never happens at the same time. - We are allocating 1GB memory on startup to each instance. - We store the location DB in a psql DB and we load it at startup. - We didn't find any errors in syslog, even at debug level. Has anyone experienced a similar issue who can suggest a possible solution? Thanks, Rogelio Perez Telnyx

6 23

Ipv6_to_ipv4
by Karlheinz Knapp 04 Sep '18

04 Sep '18

i have tested the sample config from: https://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 i found that i can register with v4 to kamailio and to the provider if i user ipv6 to kamailio and ipv4 to provider i can´t register. the packet from the provider contain a via entry which is missing from kamailio to my phone. Someone have a solution ?

1 0

event_route [tcp:closed] not executed
by Juha Heinanen 04 Sep '18

04 Sep '18

Based on comment: https://github.com/kamailio/kamailio/issues/171#issuecomment-418175840 I gave event_route [tcp:closed] a try: event_route [tcp:closed] { # Handle TCP connection close xlog("L_INFO", "$proto connection closed ($conid)\n"); } Nothing comes to syslog when I quit my baresip UA that according to wireshark closes its TCP connection after un-registration (fin - fin - ack). I have not set closed_event param, because according to README, it defaults to 1.

2 10

dispatcher probe
by volga629＠networklab.ca 04 Sep '18

04 Sep '18

Hello Everyone, Is possible access OPTIONS packet generated by dispatcher probe ? I am trying insert something like this. if(is_method("OPTIONS") && $hdr(User-Agent)=~"kamailio") { append_hf("X-Node-ID: $avp(node_id)\r\n"); } volga629

3 4

Can't connect to local MySQL server (13 "Permission denied")
by Ginhoux, Patrick 04 Sep '18

04 Sep '18

Hi, In a previous post ([SR-Users] Questions about changes between 5.0.1 and 5.0.7 <https://lists.kamailio.org/pipermail/sr-users/2018-August/102442.html> ), I reported problem to upgrade my kamailio from 5.0.1 to 5.0.7-7 level. With help from some people, it was finally fixed on my test server in my lab. Now I was expecting to do the same upgrade on a real server using these packages: kamailio-5.0.7-7.el7.x86_64.rpm kamailio-ims-5.0.7-7.el7.x86_64.rpm kamailio-mysql-5.0.7-7.el7.x86_64.rpm kamailio-perl-5.0.7-7.el7.x86_64.rpm kamailio-snmpstats-5.0.7-7.el7.x86_64.rpm But kamailio failed to connect the local MySQL server : Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: <core> [db.c:209]: db_bind_mod(): using db bind api for db_mysql Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: <core> [db.c:314]: db_do_init2(): connection 0x7fef779958f8 not found in pool Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: db_mysql [km_my_con.c:98]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/siprouter Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (13 "Permission denied") Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: htable [ht_db.c:92]: ht_db_open_con(): failed to connect to the database Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: <core> [core/sr_module.c:974]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) My "kamailio.cfg" contains the following command: modparam("htable", "db_url", "mysql://siprouter:password@localhost/siprouter") The service unit : "/usr/lib/systemd/system/kamailio.service" content is : [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=network.target mysql.service [Service] Type=simple User=kamailio Group=kamailio Environment='CFGFILE=/etc/kamailio/kamailio.cfg' Environment='SHM_MEMORY=1024' Environment='PKG_MEMORY=8' EnvironmentFile=-/etc/sysconfig/kamailio ExecStart=/usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY Restart=on-failure [Install] WantedBy=multi-user.target The "/etc/sysconfig/kamailio" content is : # # Kamailio startup options # # # Kamailio startup options # # Set to yes to enable kamailio, once configured properly. RUN_KAMAILIO=yes # User to run as USER=kamailio # Group to run as GROUP=kamailio # Amount of shared memory to allocate for the running Kamailio server (in Mb) SHM_MEMORY=1024 # Amount of per-process (package) memory to allocate for Kamailio (in Mb) PKG_MEMORY=8 Below the existing user in mysql : mysql:root:(none)> use mysql; Database changed mysql:root:mysql> select user,host from user; +-----------+-----------+ | user | host | +-----------+-----------+ | myadmin | % | | siprouter | % | | kamailio | localhost | | root | localhost | +-----------+-----------+ I have a symlink to locate the my.cnf : ln -s /myqdata/mysiprdb/data/my.cnf /etc/my.cnf And also this link to fix a htable reload problem : ln -s /myqdata/mysiprdb/data/mysiprdb.sock /var/lib/mysql/mysql.sock I don't find what the problem is this time. So if there is people that could help on the issue, thanks in advance. Cordialement. Patrick GINHOUX

3 6

load-balance to registered trunks?
by David Villasmil 03 Sep '18

03 Sep '18

Hello guys, I've got this scenario where some gateways will register on my kamailio and i need to send them calls on a load-balance fashion. is this possible? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

4 6

Dispatcher question
by Igor Olhovskiy 03 Sep '18

03 Sep '18

Hi! I’m having some issues with dispatcher and cannot get why it’s not working Idea is to use ds_next_dst right after ds_select_dst I’m having it like this modparam("dispatcher", "list_file", "/tmp/dispatcher.list") modparam("dispatcher", "flags", 2) # Enable dispather failover modparam("dispatcher", "xavp_dst", "_dispacher_dst_list_") … ds_select_dst(«1», «0»); … # Some extra check on $du here … ds_next_dst(); But get in logs WARNING: dispatcher [dispatch.c:2285]: ds_update_dst(): no xavp uri field in next destination record But contents of /tmp/dispatcher.list is 1 sip:172.28.0.12:5060 1 sip:172.28.0.19:5060 Can I somehow dump xavp_dst var to check if ds_select_dst is correct? kamailio 5.2.0-dev6 Regards, Igor

5 17

handle_ruri_alias() question or issue?
by Wilkins, Steve 02 Sep '18

02 Sep '18

Good Morning All, The following Incoming request came in => SIP Incoming Request: [[BYE sip:2406506175@20.20.20.20:5060;alias=10.10.10.5~55157~2 SIP/2.0 I called handle_ruri_alias(), and expected the destination and port to be set to 10.10.10.5:55157, but it was not. It was left 20.20.20.20:5060. Is my thinking correct or is there another way to set the destination and port to the alias? Thanks All!, -Steve

3 9

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). Any help would be massively appreciated. The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK<sip:proxy@%7bASTERISK> EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}<sip:%7bCLIENT%20IP%7d>>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}<sip:proxy@%7bPROXY%20IP%7d>> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY<sip:%7bPROXY> IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}<sip:proxy@%7bASTERISK%20INTERNAL%20IP%7d>>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}<sip:%7bPROXY%20IP%7d>> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060<sip:asterisk@%7bASTERISK%20EXTERNAL%20IP%7d:5060>> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

CentOS_7/kamailio_v5.1.x-rpms/ [Errno -1] Package does not match intended download.
by Denys Pozniak 31 Aug '18

31 Aug '18

Hello! Please check repository for Kamailio v5.1.x for CentOS7: http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… I got error: kamailio-debuginfo-5.1.5-1.el7 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not SatisfiableB --:--:-- ETA Trying other mirror. kamailio-outbound-5.1.5-1.el7. FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. kamailio-5.1.5-1.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-debuginfo-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-outbound-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. -- BR, Denys Pozniak

2 1

Re: [SR-Users] Final reply code is wrong in case of 302 and 486 busy
by Marco Capetta 31 Aug '18

31 Aug '18

Hi Daniel, did you have some time to have a look at the issue? Thank you very much Regards Marco

2 1

How to detect and Hangup a call on network lost at Caller or Callee?
by Hamid Hashmi 31 Aug '18

31 Aug '18

Dear Users Please advise how to detect a network lost at caller or callee side. I have tried timeout_socket<https://kamailio.org/docs/modules/5.1.x/modules/rtpproxy.html#rtpproxy.p.ti…> and keepalives <https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dialog.p.ka_int…> method but nothing worked for me. RTPproxy did not send timeout notification to Kamailio (Issue #1597<https://github.com/kamailio/kamailio/issues/1597>) while there is no options packet within a dialog. Is there any other method to cater such scenario. Regards Hamid R. Hashmi

5 11

Quick way to enable/disable all uac outbound registrations
by Joel Serrano 31 Aug '18

31 Aug '18

Hi, On an active/passive kamailio cluster, we use keepalived to move a virtual IP between the nodes and all signaling goes to that VIP. One of our DID providers requires us to register with every single DID we have from them in order to receive calls for such DID (NOTE: we only want to have the "active" kamailio sending outbound registrations to the carrier). My idea was to have keepalived enable all the outbound registrations when the current node changes to MASTER and disable them when it changes to BACKUP. Is there a quick "enable all" or "disable all" for uac module outbound registrations or do I have to "know in advanced" all of them and enable/disable one by one? In the docs we have: *8.4. uac.reg_disable* *Disable a remote registration record based on a filter. The command has two parameter: attribute and value. The attribute can be: l_uuid, l_username, r_username or auth_username. The value is what should be matcheg against the value of the attribute in the remote registration record.* I don't know if there is an "all" or maybe if a wildcard would work, but from the description it doesn't seem to. Any ideas? Do you guys think this is taking a correct approach? All suggestions are more than welcome!! Thanks, Joel.

3 7

Request Line alases
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, If the below Requests from 40.40.40.40(Asterisk) is received by 10.10.10.10(Kamailio) Request Line: INFO sip:3145553313@10.10.10.10:5060;alias=30.30.30.30~43508~2 SIP/2.0 Shouldn't Kamailio forward this request to 30.30.30.30? so that the '200 OK' can sent back to Asterisk for the response. (In this scenario) Would it be ok to just force a '200 OK' back to Asterisk from Kamailio? Thank you,

2 1

Delay
by Daniel W. Graham 30 Aug '18

30 Aug '18

Recently encountered a delay issue, packets arrive on system but Kamailio doesn't process right away. Here are a few details, any direction on what I can do to narrow down is appreciated. The receive queue normally sits at 0 or has random spike, when this issue occurs receive queue remains high like the below output. Version 5.1.5 ss -4 -n -l | grep 5060 udp UNCONN 321536 0 207.241.130.2:5060 *:* tcp LISTEN 0 128 207.241.130.2:5060 *:* -dan

2 2

IPv6 2 Ipv4
by Karlheinz Knapp 30 Aug '18

30 Aug '18

Hi i serach for a config to convert IPv6 to ipv4. My Provider give me only IPv4 and i have 4000 Customer which have ipv6. Anybody have expierience and a running config ?

3 2

Interesting issue with ACK back from 200 OK INVITE
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, I have had an issue for quite some time where I needed to open two UDP ports (via LISTEN) for some softphone UAC's to stay connected. Funny enough there is one Provider that works with the Single UDP. I think I finally see why I was needing two listening UDP ports. When the ACK is returning from Asterisk to Kamailio, the ACK is coming back as protocol TCP, not SIP, causing the call to drop. I finally realized this but I am not sure what would cause this and why one provider phone works but not the others. Once this is resolved, I may actually say...It works! Thank you, -Steve

1 0

Notify: stateless error reply used: I'm terribly sorry, server error occurred
by Soltanici Ilie 29 Aug '18

29 Aug '18

For some reason, Voicemail Notify messages coming from Asterisk Boxes - are not forwarded to the SIP Extensions by Kamailio. This is the SIP Packet coming to the Kamailio: 2018/08/28 10:59:19.165644 1.1.1.1:5060 -> 2.2.2.2:5060 NOTIFY sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK5f628579;rport Max-Forwards: 70 Route: <sip:2.2.2.2:5060;transport=tcp;ftag=0472e823;lr=on;nat=yes> From: "asterisk" <sip:asterisk@1.1.1.1>;tag=as1833c329 To: <sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2>;tag=0472e823 Contact: <sip:asterisk@1.1.1.1:5060> Call-ID: 93183ZjE3YmI0OGYxMDNiMzhlNzVjZjRjNTc2NTMwZGNmYzk CSeq: 103 NOTIFY User-Agent: NUACOM Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@1.1.1.1 Voice-Message: 0/0 (0/0) And this is the message which I'm receiving in the Kamailio logs: [RELAY] Error when forwarding Request ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) This is the Relay route: # Forwarding Statefully if ( ! t_relay()) { xlogl( "L_ERR" , "[RELAY] Error when forwarding Request \n " ); sl_reply_error(); } exit ;

2 2

Info: Cluecon Weekly about Kamailio, today at 17:00GMT
by Daniel-Constantin Mierla 29 Aug '18

29 Aug '18

Hello, if you haven't seen the news article on kamailio.org, today at 17:00GMT I will be part of the Cluecon Weekly online video conference to present about what's new lately in Kamailio, plans for the future as well as address questions about using Kamailio together with FreeSwitch. Anyone can join via audio or video calls, more details about at: * https://www.kamailio.org/w/2018/08/kamailio-on-cluecon-weekly-aug-29-2018/ * https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 1

Kamailio consuming UDP packets very slowly leading to high RECV - Q
by sagar malam 29 Aug '18

29 Aug '18

Hello, I am using Kamailio as a SIP proxy.So it receives SIP packet from internet and forwards it to FS servers in local network.When i execute "ss" command i see very high value in RECV-Q column.I THINK IT IS NOT NORMAL.PLEASE CORRECT ME IF I AM WRONG. ==================================================== [root@fep-1 proc]# ss -u -a -n -e | grep 5060 UNCONN 0 0 10.50.8.1:5060 *:* ino:6831630 sk:fd <-> UNCONN 0 0 10.50.7.18:5060 *:* ino:6831629 sk:fe <-> UNCONN *1183104* 0 10.50.7.254:5060 *:* ino:6831627 sk:ff <-> UNCONN *84864* 0 2607:f900:1:3::254:5060 :::* ino:6831628 sk:100 v6only:1 <-> ====================================================== Initially i thought that there is something in script which must be causing kamailio to process UDP request slower but i faced same issue with a very simple script where i simply reply with stateless 200 OK for each sip request : request_route { sl_send_reply("200","OK");exit; $avp(uuid) = $rm + "-" + $ci; ........ ......... ......... ....... Server configuration : OS : CENTOS 7 Kernel : 4.16 CPU : 5 X Intel(R) Xeon(R) CPU E5-2680 v2 @ 2.80GHz RAM : 32 GB Please help me debugging this issue.Thanks in advance

4 9

Using SEMS in IMS-ZTE Network
by Mojtaba 28 Aug '18

28 Aug '18

Hello, I am working with IMS (ZTE vendor), I want to use SBC in front of IMS network for some security policy (Hide Topology), I used SEMS as SBC in this regards, (like what we have in IMS/SEMS in kamailio by Mr Carsten Bock). When the Sebscriber want to register, The Register request received by sems and the forward to IMS(ZTE), But the "503 Service Unavailable" is received from IMS. I enable "enable_reg_caching=yes" in register.sbcprofile''s configuration file in sems. If i use kamailio, instead of SEMS, The Registration request is done successfully.In Kamailio i just forward the Register request to IMS.like this: route { if (is_method("REGISTER")) { $du = "sip:172.23.1.1:5080"; route(RELAY); } } Let me know do the issue is from myself or from IMS(ZTE)? In the fillowing, I paste all SIP Register signalling. First in kamailio, second with SEMS. ================Kamailio================== 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5e58 -> 0x4193!] UDP, length 787 E../....@.]...............^XREGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 08:20:53.114198 IP (tos 0x0, ttl 255, id 25839, offset 0, flags [none], proto UDP (17), length 685) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 657 E...d......$..............l.SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipTR9m3KdZ*3-5-16648*daei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:20:53.120142 IP (tos 0x10, ttl 64, id 48874, offset 0, flags [none], proto UDP (17), length 1137) 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5f9a -> 0xcc62!] UDP, length 1109 E..q....@.\V.............]_.REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> To: <sip:+985137054518@r-kh.com;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",uri="sip:r-kh.com;transport=UDP",response="9469852a06c01efeabf3a7a5dd1a9f3b",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Content-Length: 0 08:20:53.203871 IP (tos 0x0, ttl 255, id 25842, offset 0, flags [none], proto UDP (17), length 1118) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 1090 E..^d......p.............J.5SIP/2.0 200 OK Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipjRydzgMa*3-5-16648*daej.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Contact: <sip:+985137054518@192.168.122.1:64407;transport=UDP;rinstance=e664e178fc12691e>;expires=1800 Contact: <sip:ZTEAGCF000f425704be720000@agcf2.r-kh.com:5079;appid=2-7;UserRegInfo=002-02-00-0b80-04-172.30.48.143;BACaddr=172.30.48.143;BACport=2944>;expires=1800;+sip.instance="<urn:uuid:172.30.35.41-000>";reg-id=1 P-Associated-URI: <tel:+985137054518> P-Associated-URI: <sip:+985137054518@r-kh.com> Date: Sat, 25 Aug 2018 12:25:17 GMT Authentication-Info: nextnonce="00cfbd64b136c12a4e27b67fd28d19fb",qop=auth,rspauth="91ca956c5eb6ee9f1934e2e770750077",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001 Content-Length: 0 ============================SEMS======================== 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5e16 -> 0xe11e!] UDP, length 721 E...C.@.@.................^. REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bK~0uk0ahN;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.790414 IP (tos 0x0, ttl 255, id 35790, offset 0, flags [none], proto UDP (17), length 593) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 565 E..Q.....................=..SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bK~0uk0ahN To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipDEmH2AWT*3-5-16648*egei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 CSeq: 10 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:23:32.855067 IP (tos 0x0, ttl 64, id 17364, offset 0, flags [DF], proto UDP (17), length 1071) 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5f58 -> 0xf553!] UDP, length 1043 E../C.@.@................._X REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bKeAfKkaZa;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",uri="sip:r-kh.com;transport=UDP",response="017efb986e17506a82bb01e8eb136b90",cnonce="3764fd5eb2ea9539d61ec14196a2bf00",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.860417 IP (tos 0x0, ttl 255, id 35804, offset 0, flags [none], proto UDP (17), length 508) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 480 E...........................SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bKeAfKkaZa To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipxxsD3FR2VG8e9zxR_1dx*1-1-16648*ihad.1 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 CSeq: 10 REGISTER X-ZTE-Cause: "CSCF-1.1.BC004016-BC004288-0-0.pcscf1.r-kh.com" Retry-After: 30 Content-Length: 0 -- --Mojtaba Esfandiari.S

2 3

Parmas in lcr_gw
by Ashutosh chaubey 28 Aug '18

28 Aug '18

Hi, How to give username and password for the gateway in params column in lcr_gw? Regards, Ashutosh

2 1

regular expression in kamailio
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi All, I am trying get the 10digit number called in on using the following $var(caller) = $(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]}); I thought I could use a regular expression for the expression in re.subst,expression Thank you, -Steve

2 8

Listner problems
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi Joel, Actually, I have a strange issue. If I have a single UDP IP:Port listening, most Providers/Phones have two-way Audio/Video, however, with some Providers/Phones, I need to have two UDP IP:Port listners in order to get two-way Audio/Video. Here is the strange thing. If I enable two, the Providers/Phones that worked when I had a single listener only have one-way Audi/Video. I use Kamailio 5.1 and Asterisk 15.3 (pjsip). This behavior is so strange, Thank you, -Steve

2 2

add listen dynamically
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi All, Is it possible to add or remove "listen" dynamically? Thank you, -Steve

2 2

Struggling with RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi Florian, Thank you for your response. I checked and direct-media is off. Just as a recap, here is where I am I just can't get rtpengine to work. I have tried multiple configurations, but to no avail. Note that calls work good if rtpengine is disabled. Here is my setup => Public IP: 20.20.20.20 Private IP 10.10.10.10 flow => webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip phone rtpenngine startup (I have tried a few different startups) => rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10\!20.20.20.20 --listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 10000 -M 20000 -E kamailio => Invites: rtpengine_manage("trust-address replace-origin replace-session-connection direction=ext direction=int ICE=remove RTP/AVP"); Reply's: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); I have tried direction ext ext; and many other combinations, each producing its own incorrect behavior. Thanks again, Steve

1 0

Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hello all, I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this? I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience. Thanks all, -Steve

8 17

Send a BYE for the call
by Nicolas Breuer 23 Aug '18

23 Aug '18

Hello, Any ideas how to cut a call in progress ? How to send a forced "BYE" to all parties ? Thanks

2 2

Kamailio 5.1.4 crashes
by Abdulaziz Alghosh 23 Aug '18

23 Aug '18

Hi Everyone, I am trying to start kamailio 5.1.4 with two own modules. After /etc/init.d/kamailio start , I am getting one error of the dispatcher module and bunch of warrnings because of missed reply and failure routes. The first error before crashing is: ERROR: dispatcher [dispatcher.c:814]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Later on, when I am trying to register a user (from a soft phone) i am getting the following: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 20 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:746]: handle_sigs(): child process 6942 exited by a signal 11 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:749]: handle_sigs(): core was not generated Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [main.c:771]: handle_sigs(): terminating due to SIGCHLD Aug 22 10:02:52 sgw1a /sbin/kamailio[6959]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6958]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6957]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6956]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6955]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6954]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6953]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6950]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6941]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6944]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6948]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6951]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6946]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6952]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6943]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6949]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6947]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6938]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6937]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6940]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6939]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a kernel: [296301.271457] kamailio[6942]: segfault at 0 ip (null) sp 00007fffb3ddf368 error 14 in kamailio[400000+442000] Aug 22 10:02:52 sgw1a /sbin/kamailio[6945]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6936]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_fifo.c:594]: jsonrpc_fifo_destroy(): cannot delete the fifo (/tmp/kamailio_jsonrpc.fifo): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_sock.c:511]: jsonrpc_dgram_destroy(): cannot delete the socket (/tmp/kamailio_rpc.sock): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regarding to the first error of the dispatcher module i could not find out which calibration i can follow with its parameters to get rid of this error. Regarding the jasonrpcs and the last two erros, here are my configuration in kamailio.cfg: #/* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "fifo_mode", 0755) modparam("jsonrpcs", "fifo_group", "kamailio") modparam("jsonrpcs", "fifo_user", "kamailio") #/* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") modparam("jsonrpcs", "dgram_mode", 0755) modparam("jsonrpcs", "dgram_user", "kamailio") modparam("jsonrpcs", "dgram_group", "kamailio") I think the root cause is even beyond the dispatcher and jsonrpcs modules setting. Please check the attached where you can find my complete log file after starting Kamailio. Best regards Abdulaziz

2 4

Dispatcher - Force one end point active
by Marrold 23 Aug '18

23 Aug '18

Hi, We are currently using the dispatcher module and probing end points with OPTIONS, and making them active / inactive depending on the response: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;code=404") modparam("dispatcher", "ds_ping_from", "[REDACTED]") modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_probing_mode", 1) One of our suppliers endpoints doesn't respond to SIP OPTIONS, is it possible to force a single endpoint as active, but continue probing the rest? I've tried various settings - both globally and flags for the individual end point, but I couldn't seem to find the correct combination. Thanks Matthew

2 2

Re: [SR-Users] Fail to reload htable
by Henning Westerholt 23 Aug '18

23 Aug '18

Am Montag, 20. August 2018, 20:58:43 CEST schrieb Soltanici Ilie: > In the logs, I see these errors: > > ERROR: <core> [db.c:463]: db_use_table(): invalid parameter value > Aug 20 19:56:41 kamailio[12716]: ERROR: <core> [db.c:463]: db_use_table(): > invalid parameter value Aug 20 19:56:41 kamailio[12716]: ERROR: htable > [ht_db.c:211]: ht_db_load_table(): failed to use_table Hi, please keep the list in CC. This error is related to a database issue. Please double check if there are any issues during startup in the log. Do you see any other error there? A similar (not 100% identical) issue was discussed recently here: https://lists.kamailio.org/pipermail/sr-users/2018-August/102393.html Maybe this helps you as well. Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

4 7

Re: [SR-Users] Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 23 Aug '18

23 Aug '18

Good morning Daniel, If RTPEngine is on the same server as Kamailio (Asterisk being on another server), and RTP traffic is sent to and from RTPEngine, then the provider only needs to whitelist one IP-Address. I thought with RTPEngine that all RTP traffic would go through it and then it would pass it on to the correct destination. Is this correct? Thank you -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Daniel Tryba Sent: Thursday, August 23, 2018 4:36 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine MITRE WARNING: Do not open unexpected password-protected attachments. >>>Email originates from a non-MITRE system. Use caution.<<< On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote: > The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something? In what sense do they need whitelisting? In a common NATed solution where is no white/blacklist needed. UA gets RTP endpoints from SDP, starts sending packets to ip/port and the destination will send back packets to the source ip/port, the router/firewall will just send this to the actual UA. I have yet to find an UA that cares about where the RTP stream is coming from with regards to the SIP traffic.

2 1

SIP and RTP Proxy for multiple Servers
by sewlist＠gmail.com 23 Aug '18

23 Aug '18

Im very new to kamailio, we have some ptime issues which causes one way audio due to varies PBX vendors connecting to our class 5 softswitches I would like to use Kamailio as the SIP/RTP proxy in front of our class 5 soft switches I followed the example of freeswitch and kamailio. However the config is still hard for me below setup works, however RTP not being proxied, Looks like its its not, because im not using NAT, so RTP goes direct https://freeswitch.org/confluence/display/FREESWITCH/Kamailio+basic+setup+a… I would like to to ensure RTP proxy is always used(possible re-framing ptime) and we have multiple soft switches Is it possible to setup kamailio.cfg so my customers can point the endpoint to kamailio and domain names contains the server we want to use to register with? eg, softswitch 1 = 1.1.1.2 softswitch 2 = 1.1.1.3 softswitch 3 = 1.1.1.4 etc etc my current config #!KAMAILIO #!define IPADDRESS "1.1.1.1" #!define SWITCH_IPADDRESS "1.1.1.2" #!define FLAG_FROM_SWITCH 1 #!define FLAG_FROM_USER 2 # ------------------ module loading ---------------------------------- loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "path.so" loadmodule "rtpproxy.so" # ----------------- setting module-specific parameters --------------- modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") # ------------------------- request routing logic ------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif # main routing logic route { # per request initial checks route(SANITY_CHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } route(CHECK_SOURCE_IP); ################################## ### HANDLE SEQUENTIAL REQUESTS ### route(WITHINDLG); ############################### ### HANDLE INITIAL REQUESTS ### t_check_trans(); if (is_method("INVITE|REFER")) { record_route(); } if (is_method("REGISTER")) { add_path(); } if (isflagset(FLAG_FROM_SWITCH)) { # don't send INVITE from SWITCH back to SWITCH, set reply route to handle NAT and forward them along t_on_reply("EXTERNAL_REPLY"); } else { # set destination to your SWITCH $du = "sip:1.1.1.2:5060"; } route(RELAY); } route[SANITY_CHECK] { if (!sanity_check()) { #xlog("L_WARN", "$ci|end|message is insane"); exit; } if (!mf_process_maxfwd_header("10")) { #xlog("L_WARN", "$ci|end|too much hops, not enough barley"); send_reply("483", "Too Many Hops"); exit; } if ($ua == "friendly-scanner" || $ua == "sundayddr" || $ua =~ "sipcli" ) { #xlog("L_WARN", "$ci|end|dropping message with user-agent $ua"); exit; } if ($si == IPADDRESS) { #xlog("L_WARN", "$ci|end|dropping message"); exit; } } route[CHECK_SOURCE_IP] { if ($si == SWITCH_IPADDRESS) { setflag(FLAG_FROM_SWITCH); } else { setflag(FLAG_FROM_USER); } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(RELAY); } else { if (is_method("NOTIFY")) { route(RELAY); } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard #xlog("ACK without matching transaction ... ignore and discard"); exit; } } sl_send_reply("404","Not here"); } exit; } } onreply_route[EXTERNAL_REPLY] { route(NAT_TEST_AND_CORRECT); } route[NAT_TEST_AND_CORRECT] { if (nat_uac_test("3")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { #setbflag(FLB_NATB); route(RTPPROXY); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif route(RTPPROXY); /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; # if (!t_relay()) { # sl_reply_error(); # } # exit; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { #unforce_rtp_proxy(); force_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; }

2 1

RTPEngine wow's
by Wilkins, Steve 22 Aug '18

22 Aug '18

Hello all, I am still trying to get RTPEngine to work. At this point, when I make a call, I do see the offer in the rtpengine log, I also see the update of the SDP (c) changing it to 20.20.20.20. However, after that, there are no logs other than a bunch of "timer run time = 0.nnnnnnn sec", and the delete when I hang up the call. There is no RTP traffic what so ever in the rtpengine log (however, the call is completely successful, other than the RTP traffic not going through RTPEngine). Here is a breakdown of my configs (I borrowed this format from Maxim Fedorov because I thought it was very concise) Starting rtpengine => rtpengine --interface=internal/21.21.21.21--interface=external/21.21.21.21\!20.20.20.20 --listen-ng=127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1 RTPEngine Public IP: 20.20.20.20 RTPEngine Private IP: 21.21.21.21 kamailio.cfg modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12221") listen=tcp: 21.21.21.21:5060 advertise 20.20.20.20:5060 listen=udp: 21.21.21.21:5060 advertise 20.20.20.20:5060 rtpengine_manage call rtpengine_manage("trust-address replace-origin replace-session-connection external external media-address=20.20.20.20"); (I have placed this rtpengine_manage call at the beginning of all routes). I have tried many variations of rtpengine_manage parameters, and placement of the call itself, but I never see RTP traffic in the rtpengine log. Thank you to all who have looked at this, -Steve

1 0

Error at startup
by Nicolas Breuer 22 Aug '18

22 Aug '18

Hello, How can i clear this error ? Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:228]: pv_table_add(): pvar [rtpstat] already exists Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:513]: register_pvars_mod(): failed to register pseudo-variable <rtpstat> for module rtpproxy

2 2

Releasing Kamailio v5.1.5
by Daniel-Constantin Mierla 22 Aug '18

22 Aug '18

Hello, with the summer holidays in the northern hemisphere approaching the end, I am considering to release v5.1.5 out of Kamailio git branch 5.1 -- there were several fixes done since 5.1.4 and it is time to package another minor release. The propose date is next week on Wednesday, August 22, or slightly after in case there are unexpected events delays it. Should there be any issues you are aware of and not yet reported on the bug tracker, do it asap in order to have a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

1 2

Encoding mechanism in UAC Module
by Duarte Rocha 21 Aug '18

21 Aug '18

Greetings, Is there a way to choose the encoding algorithm that builds the vsf and vst parameters in the Record-Route? What is the default method that Kamailio uses for this ? Thanks in advance, Duarte Rocha

2 1

invalid version 8 for table aliases found, expected 9
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Hi everyone, i am trying to start kamailio 5.1.4 after the migration from Kamailio 3.0.3 but getting newly the following error: ERROR: <core> [db.c:450]: db_check_table_version(): invalid version 8 for table aliases found, expected 9 (check table structure and table "version") Allegedlly, this error is because of a wrong version of "aliases" table. I followed the Upgrade process starting from 3.0.3 till 5.1.x and I altered the structure of my aliase table accordinglly. The last new version of aliases is 8 as it is described at the upgrade from 4.2.x to 4.3.0 Even though, the error here says that the newest version is 9: May someone help me to find out if the aliases table's structure was renewed in between ?? Is there any modification in the data types? Thanks in advance Abdulaziz

3 4

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Greetings, i am at the final stages of emigrating from kamailio 3.0.3 to 5.1.0 including our own module. Unluckily, after writing the command " kamctl start", the following error is popping up: * PID file /var/run/kamailio.pid does not exist -- Kamailio start failed* Thus, I used the command " kamailio -M 8 -E -e -dd " to get more debugging Info and the output is detailing the following: 0(10334) ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on '192.168.1.207' (4) 0(10334) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(10334) ERROR: ABDUL_functions [ABDUL_chk.c:1069]: init_db(): failed to connect database URL=mysql://ABDUL:ABDUL@192.168.1.207/ABDUL 0(10334) ERROR: ABDUL_functions [ABDUL_functions_mod.c:296]: mod_init(): Module ABDUL_functions not correktly initialized, pid[10334] 0(10334) ERROR: <core> [core/sr_module.c:991]: init_mod(): Error while initializing module ABDUL_functions (//lib64/kamailio/modules/ABDUL_functions.so) Seemingly the error relates to old databases url ( "URL=mysql:// ABDUL:ABDUL@192.168.1.207/ABDUL" ) inside an old Kamailio.cfg file from my 3.0.3 package which I put under another different directory. This old Kamailio.cfg file #includes my own module, so he is exporting this url. Consequently, my module is trying to connect again to this databases. I do not know why both kamailio.cfg (new: /etc/kamailio/kamailio.cfg and old one out of /etc/kamailio) are both seen for kamctl or (/sbin/kamailio). I know that my new url inside the new /etc/kamailio/kamailio.cfg is read and connected but again not sure why my hidden old kamailio.cfg is still being read as well. the attached shows my init file for "SUSE Linux Enterprise Server 11 (x86_64)" and my used kamailio *default *file under /etc/default. Thanks in advance and best regards Abdulaziz

2 2

help
by Pravin . 21 Aug '18

21 Aug '18

Hello Team, I want to install Kamailio SIP server on centOS server...do we need to download kamailio ISO file and install on server..pls provide the installation procedure/guidelines. Regards Pravin

6 14

Dialplan priority
by igor.potjevlesch＠gmail.com 21 Aug '18

21 Aug '18

Hello, I'm not sure to understand in which situation "pr" value is used. For example: if I have two identical rules except pr value: 1 with 1 and the second with 2, then the rule with pr 2 is never used. Someone can help me to understand? Thank you. The finality I'd like to accomplish is having some kind of load balacing between two values into "repl_exp" with the same inputs conditions. Thank you. Regards, Igor.

2 2

Call Transfer Scenario Using Freeswitch
by Sunil More 21 Aug '18

21 Aug '18

Hi All, I am using kamailio and freeswitch in my setup. While all users are registered on Kamailio, calls are routed to Freeswitch back to kamailio and then upstream providers. I can make user to user calls but in cases of transfer I am stuck. I am using a Polycom IP phone to test this scenario. While I initiate a attended transfer and click on transfer all the legs hangup. When I register all users on Freeswitch this works well. I am missing something on Kamailio side. Please point me in the right direction. Thanking You, Sunil More Ph : 9503338275

2 1

Enabling tcp_write_buf
by Gholamreza Sabery 21 Aug '18

21 Aug '18

Hi, In the documentation of "tcp_wq_max" option it is written that : > It has no effect if *tcp_write_buf *is not enabled. > > The interesting fact is that, there is not description of what it is and how one can enable it. There is even a question about it in the mailing lists: https://lists.kamailio.org//pipermail/sr-dev/2011-June/011521.html It seems that it is an inaccuracy in the documentation. If there is such an option how to enable it? Can anyone tell? Regards

3 2

Best practice/routing block for TCP connection errors to UAC?
by Anthony Alba 21 Aug '18

21 Aug '18

HI SR-Users Scenario: clients registered by TCP/TLS dropping off the network, kamailio stil thinks the client is registered and relays the INTIVE. The error occurs at the TCP layer so no failure route is called. ERROR: <core> [core/tcp_main.c:2690]: tcpconn_1st_send(): connect 127.0.0.1:48338 failed (RST) Connection refused ERROR: <core> [core/tcp_main.c:2700]: tcpconn_1st_send(): 127.0.0.1:48338: connect & send for 0x7f5494e906c0 failed: Connection refused (111) ERROR: tm [../../core/forward.h:251]: msg_send_buffer(): tcp_send failed WARNING: tm [t_fwd.c:1567]: t_send_branch(): sending request on branch 0 failed ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) I want to send the caller instead a graceful failure route type message as if the callee is not registered (404/486/603/408 type) but don't understand how to hook a routing block for TCP errors. Anthony

1 0

Fail to reload htable
by Soltanici Ilie 20 Aug '18

20 Aug '18

I'm trying to reload a hash table but receiving an error: error: 500 - Mtree reload failed [root@kamailio]# /usr/local/sbin/kamcmd htable.listTables { name: auth dbtable: dbmode: 0 expire: 0 updateexpire: 1 size: 256 dmqreplicate: 0 } [root@kamailio]# /usr/local/sbin/kamcmd htable.dump auth { entry: 69 size: 1 slot: { { name: username::8.8.8.8 value: 0 type: int } } } [root@kamailio]# /usr/local/sbin/kamcmd htable.reload auth error: 500 - Mtree reload failed [root@kamailio]# /usr/local/sbin/kamailio -v version: kamailio 5.1.4 (x86_64/linux) e8b2bf

2 1

Kamailio 5.1 Nathelper sipping on different transport protocols.
by Володимир Іванець 20 Aug '18

20 Aug '18

Hello everyone! I have system with Kamailio v5.1 which needs to send OPTIONS packets to all active clients. Here is full Nathelper module configuration: *modparam("nathelper", "received_avp", "$avp(s:received)")* *modparam("nathelper", "natping_interval", 30)* *modparam("nathelper", "ping_nated_only", 0)* *modparam("nathelper", "sipping_bflag", FLB_SIPPING)* *modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org <sip%3Apinger(a)kamailio.org>")* It works fine if client is connected over UDP. But as soon as I will switch to TCP or register with client that uses WS, Kamailio stops generating OPTIONS. Does anyone know if it is correct behavior or something is wrong? Thanks a lot!

3 3

remove_hf function in INVITE message
by Mojtaba 20 Aug '18

20 Aug '18

Today i encounter with strange issue in SIP signalling in Kamailio, In INVITE message, I just change Contact Header and relay message to destination, But the message is like this: 192.168.122.245.4080 > 192.168.122.174.5060: [bad udp cksum 0x7b5e -> 0x3a4d!] SIP, length: 1104 INVITE sip:8000@r-kh.com;transport=UDP SIP/2.0 Record-Route: <sip:192.168.122.245:4080;lr;ftag=8e47e95e;nat=yes> Via: SIP/2.0/UDP 192.168.122.245:4080;branch=z9hG4bK14d7.dc974309aba269b24213204d3ea085e2.0 Via: SIP/2.0/UDP 192.168.122.1:36853;rport=36853;branch=z9hG4bK-d8754z-423b431a3ebc8a22-1---d8754z- Max-Forwards: 69 sip:7000@192.168.122.1:36853;transport=UDP;alias=192.168.122.1~36853~1To: <sip:8000@r-kh.com;transport=UDP> From: <sip:7000@r-kh.com;transport=UDP>;tag=8e47e95e Call-ID: YTZmYTJmYzg0ZDNiMjM2OGUxZGYwNzg4OWY1ZmIxYzc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 <SDP Body> The code in Kamailio is: #!ifdef WITH_SBC_NASIM #if the Invite request is MO, forward it to Asterisk . if (is_method("INVITE") && $si!="192.168.122.245") { xlog("L_INFO", "The Invite message from $si:$sp, Forwart to Asterisk. <192.168.122.174:5060>.\n"); remove_hf("Contact"); append_hf("Contact:sip:$fU@sbc.r-kh.com\r\n"); $du = "sip:192.168.122.174:5060"; } #!endif Be notice that, i have the same procedure in REGISTER message, But it is work correct, if(is_method("REGISTER")) { xlog("L_INFO", "forward register to Asterisk\n"); remove_hf("Contact"); append_hf("Contact:sip:$fU@sbc.r-kh.com\r\n"); rewriteuri("sip:r-kh.com"); $du = "sip:192.168.122.174:5060"; route(RELAY); } Let me know what is the problem? -- --Mojtaba Esfandiari.S

2 2

Upgrade from Kamailio 4.4>5
by Nicolas Breuer 20 Aug '18

20 Aug '18

Hello, After upgrading, i cannot start anymore the service. Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/pvapi.c:828]: pv_parse_spec2(): error searching pvar "NULL" Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/pvapi.c:1032]: pv_parse_spec2(): wrong char [L/76] in [$NULL] at [4 (0)] Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/rvalue.c:2551]: mk_rval_expr_v(): memory allocation failure Any ideas ?

2 2

JSONRPC-S and async RPC commands
by Juha Heinanen 20 Aug '18

20 Aug '18

JSONRPC-S module README has: 2. Limitations This module does not implement asynchronous RPC commands. It is unlikely that asynchronous RPC commands will be executed from an JSON-RPC over HTTP client. Does this mean that, for example, tm.t_uac_wait command cannot be executed using JSON-RPC over HTTP? If so, why "it is unlikely that asynchronous RPC commands will be executed from an JSON-RPC over HTTP client", since those commands are currently commonly executed from XMLRPC client using xmlrpc module? It would be nice if JSONRPC-S module could be used as replacement of xmlrpc module. -- Juha

2 8

domain-based routing
by David Villasmil 19 Aug '18

19 Aug '18

Hello all, I need to set up a multi-domain kamailio, i think in terms of accepting invites to different domains is pretty simple by using the "alias" parameter, that much i have clear. My doubt comes from the fact that I need to route outgoing calls also based on the domain. As of today, i have single-domain kamailios using dispatcher to load-balance and failover said calls. I think i should be able to use dispatcher here as well, separating the dispatcher list by GID, so that Domain1->gid:1 Domain2->gid: 2 Domain3->gid: 3 and so on... I don't have registrations on this server and none are expected. So i'm thinking i will received the invites like: sip:12345@domain1.com sip:12345@domain1.com sip:12345@domain1.com and so on... How would you guys go about associating the domain with the gid list in dispatcher? Thanks a lot for your help! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Re: [SR-Users] Mysql 5.7 vs 8.0
by Henning Westerholt 17 Aug '18

17 Aug '18

Am Freitag, 17. August 2018, 16:37:23 CEST schrieb Wilkins, Steve: > Yes, I did read that post, thank you for that; my problem is that installing > Kamailio is one of the last things I do in a script so I was hoping to find > a way to tell Kamailio to point to a different version. I guess I will > just need to go another route. Hi Steve, (add sr-users) kamailio by itself will not force the installation of a mysql version. I think this is done from the distribution package management and from the respective kamailio package. There are ways to adapt the installed dependencies during this installation, but you need to investigate it, as its depend on your linux distribution. Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

1 0

Mysql 5.7 vs 8.0
by Wilkins, Steve 17 Aug '18

17 Aug '18

Hi All, I would like Kamailio to use MySQL 5.7, however when Kamailio installs and I say I want to use MySQL, it installs 8.0. Can I someway direct Kamailio to install version 5.7? Thank you, -Steve

2 1

Registered User limit
by Philip Fleck 17 Aug '18

17 Aug '18

Is there a limit to the number of users which can be registered on Kamailio at any one time, or a configuration file which sets this. I have just installed Kamailio in a lab environment for testing purposes but I seem to only be able to register 1024 users, all remaining Register requests are ignored. I am not using a database to authenticate users or anything, just want to be able to register upto 10,000 users. regards Philip

6 6

Bogus event 8 in state 1 for dlg
by Phillman25 Kyriacou 17 Aug '18

17 Aug '18

Hi! I keep receiving the below logs on Kamailio: CRITICAL: dialog [dlg_hash.c:955]: log_next_state_dlg(): bogus event 8 in state 1 for dlg 0x7f59862cd440 [3685:10496] I understand that state 1 indicates an unconfirmed dialog, can anyone tell me what bogus event 8 is? The following is the call flow: User A ===> Kamailio ==> Provider >From the PCAP trace an INVITE is sent to the provider, the provider answers with 100 Trying, 180 Ringing and immediately an OPTIONS message with the same call id Kamailio Provider INVITE =================> 100 Trying <================= 180 Ringing <================ OPTIONS <================ Im assuming bogus event 8 is the OPTIONS message? Is an OPTIONS message normal in this case? Thanking you very much in advance for your help! Regards Phillip

3 4

register with Kamailio
by Pravin . 16 Aug '18

16 Aug '18

Hello, I have been registered with Kamalio with pravin(a)bolindia.com id ..its not working with login credential registered with kamailio. Pls suggest Regards, Pravin bolindia Networks Pvt Ltd.

2 1

Kamailio as a P-CSCF user
by serkan celik 16 Aug '18

16 Aug '18

Hi all, I want to use Kamailio as a P-CSCF on ubuntu 18. Kamailio will be p-cscf and will communicate scscf in another open-ims box. How can I implement this feature. I have already looked following link. However it did not help me since I could not see any example pcscf configuration files after installation https://www.kamailio.org/wiki/tutorials/ims/installation-howto Regards, Serkan

3 11

how to enable tcp_write_buf?
by Juha Heinanen 16 Aug '18

16 Aug '18

I tried to issue mtree.list command using xmlrpc query that would result in large response, and got error Aug 15 18:12:07 char /usr/bin/sip-proxy[10348]: ERROR: <core> [core/tcp_main.c:618]: _wbufq_add(): (221375 bytes): write queue full or timeout (0, total 0, last write 55534052 s ago) Aug 15 18:12:07 char /usr/bin/sip-proxy[10348]: ERROR: xmlrpc [xmlrpc.c:811]: send_reply(): Error while sending reply Aug 15 18:12:07 char /usr/bin/sip-proxy[10367]: ERROR: <core> [core/tcp_main.c:3551]: handle_ser_child(): received CON_ERROR for 0x7f112ad22be0 (id 5), refcnt 3, flags 0x4018 I though that perhaps I could get rid of the error by increasing value of core var tcp_conn_wq_max. Description of the var tells: Maximum bytes queued for write allowed per connection. Attempting to queue more bytes would result in an error and in the connection being closed (too slow). If tcp_write_buf is not enabled, it has no effect. What is tcp_write_buf and how to enable it? I could not find anything about it in core cookbook. -- Juha

3 7

Issue with RTPEngine(6.5.0.0+0~mr6.5.0.0)
by Mojtaba 16 Aug '18

16 Aug '18

Hello, Before i used RTPEngine (6.4.0.0+0~mr6.4.0.0) without any problem, Nowday i install RTPEngine from git. I notice that it's varsion is changed to 6.5.0.0+0~mr6.5.0.0. When i want install ngcp-rtpengine-recording-daemon, I have this issue: Failed to start ngcp-rtpengine-recording-daemon.service: Unit rpcbind.socket failed to load: No such file or directory. invoke-rc.d: initscript ngcp-rtpengine-recording-daemon, action "start" failed. dpkg: error processing package ngcp-rtpengine-recording-daemon (--install): subprocess installed post-installation script returned error exit status 6 Processing triggers for systemd (215-17+deb8u7) ... Errors were encountered while processing: ngcp-rtpengine-recording-daemon Please let me what is this issue? -- --Mojtaba Esfandiari.S

3 4

kamailio rtp proxy set not working
by ANOOP V M 16 Aug '18

16 Aug '18

I have installed kamailio 5.1.4 version in ubuntu server. I need to route RTP packets through kamailio with SIP packets. I will explain the way i have set up everything. I have installed rtpproxy by apt-get install rtpproxy. i have edited "/etc/default/rtpproxy" file with the line 'EXTRA_OPTS="-F -s udp:127.0.0.1:7722 -l 34.201.122.249 -d DBUG:LOG_LOCAL0"' i have edited "/etc/init.d/rtpproxy" file as follows, DAEMON=/usr/bin/$NAME PIDFILE="/run/$NAME/$NAME.pid I have provided "#!define WITH_NAT" and "modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")" in my cfg file. started rtpproxy "service rtpproxy start" when i try "netstat -tulpn | grep rtp" it shows udp 0 0 127.0.0.1:7722 0.0.0.0:* 1437/rtpproxy So i think the rtpproxy is currently running fine. Now we can go to kamailio part. I have gone through the document " https://www.kamailio.org/docs/modules/5.0.x/modules/rtpproxy.html" to set up rtpproxy in the cfg file. checking the rtp ports "kamcmd rtpproxy.list" showing { setid: 0 url: udp:127.0.0.1:7722 index: 0 disabled: 0 weight: 1 recheck: 0 } I have enabled rtp proxy using "kamcmd rtpproxy.enable udp:127.0.0.1:7722 1". So i think ports are ready for rtp proxy. Now i will explain the call flow. I have enabled log using L_BUG. when i try a call from hangout number through my twilio trunk to my asterisk freepbx extension, set_rtp_proxy_set("1"); giving true value. But rtpproxy_offer(); giving -1 as result. syslog shows some errors like below, Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: DBUG:handle_command: received command "1934_4 Uc0,8,3,111,9,101 52001b5b71435b1c5392b39f2f6c975b@asterisk_ip:5060 asterisk_ip 14540 as4083c85f;1" Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: INFO:handle_command: new session 52001b5b71435b1c5392b39f2f6c975b@asterisk_ip:5060, tag as4083c85f;1 requested, type strong Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: ERR:create_twinlistener: can't bind to the IPv4 port 47926: Cannot assign requested address Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: ERR:handle_command: can't create listener Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: DBUG:doreply: sending reply "1934_4 E10#012" And also my header is not changed. As a result actuall rtp is going end to end between asterisk and twilio not through kamailio. Please help me with the details.

3 4

Very strange reinvite behaviour
by Alex Balashov 15 Aug '18

15 Aug '18

Hi, We have a scenario like this: ITSP -----> Kamailio -----> Endpoint (UDP) (TLS) So, TLS is only being used on the last hop, and the upstream interaction with the ITSP is plain-old UDP. Kamailio has the following listeners: listen=udp:1.1.1.1:5060 listen=udp:1.1.1.2:5060 listen=tcp:10.0.0.1:5060 listen=tls:1.1.1.1:5061 At some point, 'Endpoint' sends a reinvite which has the following RURI: INVITE sip:1.1.1.1:5061 SIP/2.0 This is clearly improper, and caused a loop that led to the rtpengine SDP loop issue I previously reported in another thread. So, in an effort to stop this, I added the following: if(has_totag()) { if(loose_route()) { ... if(is_method("INVITE")) { xlog("L_INFO", "[R-MAIN:$ci] Re-INVITE received from $si:$sp to RURI $ru\n"); xlog("L_INFO", "[R-MAIN:$ci] Reinvite body: $mb\n"); if(!is_method("ACK") && uri == myself) { sl_send_reply("400", "Bad Request"); exit; } } } But it doesn't work. It appears that the '400 Bad Request' rejection never happens, presumably because the this domain doesn't match 'myself'. Another perplexing mystery: the log message containing the reinvite's '$ru' does not show a RURI of 'sip:1.1.1.1:5061', but rather the remote target in the initial inbound INVITE, which we also logged: Contact: "Anonymous" <sip:Restricted@3.3.3.3:5060> The log message says: [R-MAIN:...] Re-INVITE received from 4.4.4.4:5060 to RURI sip:Restricted@3.3.3.3:5060;lr. Note a subtle detail here: the ';lr' parameter is present, which is an attribute of the Record-Route inserted by the sending ITSP (3.3.3.3). It's at the bottom of the Route set, of course, below Kamailio's two RRs (inserted for the ingress UDP interface and the egress TLS interface): Route: <sip:Restricted@3.3.3.3:5060;lr> This leads to two questions whose causes seem to be related: 1. Why does Kamailio think the request URI of this re-invite is something other than what $mb reveals it to be? 2. Is that, presumably, why it does not match 'myself'? 3. Why would Kamailio think it is actually set to the far-end Record-Route URI? This is version: kamailio 4.4.5 (x86_64/linux) d48094. Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 9

rtpproxy or rtpengine with Kamailio and Asterisk
by Wilkins, Steve 14 Aug '18

14 Aug '18

Good Morning everyone, Has anyone ever gotten rtpproxy/rtpengine to work with Kamailio and Asterisk? I have tried both, and even though I see no errors with either, and when I make calls I see rtpproxy/rtpengine traffic, I am still seeing media traffic go directly to and from Asterisk. I just can't get the media packets to not go directly from Asterisk to the Callee and visa-versa. Any ideas on where I am messing up in my kamalio.cfg file? Thank you!, -Steve

3 2

Migrate from RtpProxy to Rtpengine
by Nicolas Breuer 14 Aug '18

14 Aug '18

Hello, I migrated from Rtp_proxy module to Rtp_Engine. Everythings works fine apart the rewriting of the SDP description in Kamailio subst_body('/(^s=.*)/s=Test of a Sip call.\r/') Any ideas if this is supposed to be supported?

4 11

Can Kamailio interface with an asterisk server based on the fqdn of a call?
by Wilkins, Steve 14 Aug '18

14 Aug '18

Hi All, Is it possible for Kamailio to interface with a particular Asterisk Server based on the FQDN of a caller? I would like to pass calls received by Kamailio through to different Asterisk Servers based on the FQDN of the caller. I have used Load Balancing before, but I want to select which Asterisk Server the call is directed to. Thank you,

3 2

Rtpengine?
by Wilkins, Steve 14 Aug '18

14 Aug '18

HI All, I am not sure if I understand it correctly but I thought that I could use rtpengine to redirect media packets. My current SIP flow is => Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is Asterisk<=>softphone. But I don't want Media to flow this way. That is, I do not want Asterisk and the Softphone to be aware of each other. I would like the Media to go through Kamailio just like SIP packets do. I installed rtpengine and it starts and runs with no errors. I even see traffic with the call is made but media traffic still flows from softphone<=>asterisk. Any ideas? Thank you,

2 3

Upgrade from 4.4 to 5.1
by Nicolas Breuer 13 Aug '18

13 Aug '18

Hello, I want to upgrade my both Kamailio from 4.4 to 5.1 I have to upgrade the SQL structure before running the 5.1 release. But my SQL servers are synchronized - Kamailio 4.4 can works with the 5.1 SQL structure ?

2 4

Re: [SR-Users] problem with compiling kamailio version 5.1.4 including a private Module
by Abdulaziz Alghosh 13 Aug '18

13 Aug '18

Hi Henning, thanks again. I found meanwhile the root cause. It caused as you said by the old logic of the make file. I used instead the new Formula of the tm module’s make file (as i remember). Consequently, it did not bring the compilation through those unwanted libraries. I will spot the details later when i am back in office next Monday. BR Abdulaziz On Thu 9. Aug 2018 at 21:19 Henning Westerholt <hw(a)kamailio.org> wrote: > Am Donnerstag, 9. August 2018, 14:27:44 CEST schrieb Abdulaziz Alghosh: > > thanks a lot for your response and explanation. strutils.[c,h] and > > strcommon.[c,h] are almost the same. So I modified my included header > from > > stcommon[] to strutils as you mentioned. > > Unfortunately, I surfed sequentially Kamailio version upgrades. Even > > though, i did not found such hints about this modification. So I have > some > > doubts the called headers and codes from our own module. Hence, I am > asking > > for yor advice. > > Hello Abdul, > > lets move this discussion to the sr-dev list, as its a development related > topic. > > > What I am trying to do now is compiliing our module separately under it > is > > own directory so I gain the Shared Object file (.so). Somehow our Module > is > > requring cds, presence and xcap libraries. But the strange thing is that > > headers from these libraries are #including other headers assuming these > > included header files are under the same library's directory (but in fact > > most of the header files are under ../src/core/). For example : > > Do you placed your own module in the src/modules directory as well? If > not, I > would suggest to place it there. Its hard to find the issue just with the > logs, there are also many errors here. > > My suggestion would be to go from your own module and try to compile a > single > file with gcc (for the actual command you can call "make quiet=0" and copy- > paste it). I would try to investigate one error a time, in the end they > should > be all mostly related. > > There were also some changes in the Makefile logic, maybe you should > compare > your Makefile to one of the official modules. > > > Making in cds > > CC (gcc) [L libser_cds.so.0.1] cds.o > > In file included from cds.c:2: > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:21: > > warning: mem/mem.h: No such file or directory > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:25: > > warning: mem/shm_mem.h: No such file or directory > > In file included from cds.c:3: > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: > > warning: locking.h: No such file or directory > > In file included from cds.c:4: > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20: > > error: dprint.h: No such file or directory > > make[2]: *** [cds.o] Error 1 > > make[1]: [cds] Error 2 (ignored) > > > > > > Those libraries are delivered with version 5.1.4 itself so i do not think > > it is a good idea to modify the paths of the included header files to > > compile my module. because most probably these libraries are needed by > > standard default modules of kamailio. > > > > I hope this is not too much for you to inspect. > > Note: The attached has the log files ( resulting from make all and > install > > ) . > > > > Your kind response is highly appreciated > > BR > > Abdulaziz > > > > On Thu, Aug 9, 2018 at 11:31 AM, Abdulaziz Alghosh <aziz647(a)gmail.com> > > > > wrote: > > > Hello Henning, > > > > > > thanks a lot for your response and explanation. strutils.[c,h] and > > > strcommon.[c,h] are almost the same. So I modified my included header > from > > > stcommon[] to strutils as you mentioned. > > > Unfortunately, I surfed sequentially Kamailio version upgrades. Even > > > though, i did not found such hints about this modification. So I have > some > > > doubts the called headers and codes from our own module. Hence, I am > > > asking > > > for yor advice. > > > > > > What I am trying to do now is compiliing our module separately under > it is > > > own directory so I gain the Shared Object file (.so). Somehow our > Module > > > is > > > requring cds, presence and xcap libraries. But the strange thing is > that > > > headers from these libraries are #including other headers assuming > these > > > included header files are under the same library's directory (but in > fact > > > most of the header files are under ../src/core/). For example : > > > > > > Making in cds > > > CC (gcc) [L libser_cds.so.0.1] cds.o > > > In file included from cds.c:2: > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:2 > > > 1: warning: mem/mem.h: No such file or directory > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:2 > > > 5: warning: mem/shm_mem.h: No such file or directory > > > In file included from cds.c:3: > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: > > > warning: locking.h: No such file or directory > > > In file included from cds.c:4: > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20 > > > : > > > error: dprint.h: No such file or directory > > > make[2]: *** [cds.o] Error 1 > > > make[1]: [cds] Error 2 (ignored) > > > > > > > > > Those libraries are delivered with version 5.1.4 itself so i do not > think > > > it is a good idea to modify the paths of the included header files to > > > compile my module. because most probably these libraries are needed by > > > standard default modules of kamailio. > > > > > > I hope this is not too much for you to inspect. > > > Note: The attached has the log files ( resulting from make all and > install > > > ) . > > > > > > Your kind response is highly appreciated > > > Abdulaziz > > > > > > > > > On Thu, Aug 9, 2018 at 8:11 AM, Henning Westerholt <hw(a)kamailio.org> > > > > > > wrote: > > >> Am Mittwoch, 8. August 2018, 17:47:07 CEST schrieb Abdulaziz Alghosh: > > >> > i am somehow newbie with Kamailio and trying to migrate from > kamailio > > >> > version 3.0.3 to 5.1.4. Former fellows had developed own modules > which > > >> > needed "strcommon.h" and "strcommon.c". > > >> > > > >> > These last two files are not delievered with version 5.1.4. but I > > >> > copied > > >> > them under ../src/core/ thinking it would be beneficial. > Unfortunately, > > >> > after make prefix / (and including our own module) all, it seem that > > >> > several dependencies exist. Especially from header files under > > >> > ..src/lib/xcap/ and ../src/lib/presence > > >> > > > >> > Can somebody tell me how "strcommon.h" and "strcommon.c" where > > >> > > >> replaced in > > >> > > >> > version 5.1.4 ? > > >> > > >> Hello Abdul, > > >> > > >> the functions in strcommon.[c,h] were moved into > src/core/strutils.[c,h] > > >> in > > >> releases starting from 5.0. I'd suggest to have a look to the file to > > >> verify > > >> if there are no other changes. > > Best regards, > > -- > Henning Westerholt > https://skalatan.de/blog/ >

1 1

Kamailio IMS with TLS
by Amar Tinawi 13 Aug '18

13 Aug '18

Hello i'm running tls conf over PCSCF, register and audio calls works fine, but when it comes to video calls it's dropped when it comes to video second-invite it worked when the call is initiated from the beginning as video call should i enable tls also on icscf and scscf ? or the video is not supported over tls ?

1 0

app_lua and lua_run parameter count
by Asgaroth 13 Aug '18

13 Aug '18

Hi All, Quick question with regards the lua_run function in app_lua module. The documentation for v5.0.x states: "Execute the Lua function 'func' giving params as parameters. There can be up to 3 string parameters. The function must exist in the script loaded at startup via parameter 'load'. Parameters can be strings with pseudo-variables that are evaluated at runtime." Does this mean that lua_run can only have 3 parameters, or, does it mean that the function within the lua script can have up to 3 parameters? I have a function within lua that currently needs 3 parameters, so I'm hoping I can do the following: lua_run("some_function", "param_1", "param_2", "param_3") However, I'm not sure, from the documentation, if I can do this. On a side note, but related, can I pass defined variables/constants as parameters to lua_run too, for example, something like: #!define VAR_1 1234 #!define VAR_2 hello lua_run("some_function", "$rd", VAR_1, VAR_2) Some clarification would be greatly appreciated. Thanks for reading.

2 2

Lua KEMI crash
by Ivaylo Markov 13 Aug '18

13 Aug '18

Hello, I have Kamailio 5.1.4 (from deb.kamailio.org) running on a Debian 9 AMD64 VM with 1 CPU and 1G of RAM on ESXi. For KEMI I am using the liblua5.1-0:amd64 package, version 5.1.5-8.1+b2. The configuration is rather simple - dispatcher module reading the gateways from a file and forwarding everything to them based on a custom variable. There are 2 gateways in the file being pinged with OPTIONS packets every 10 seconds. Only a single SIP user is connecting to the machine (besides the occasional scanner). I can post the config if necessary, but I don't think that is the issue (see below). The crash has happened twice already - once on 2018-08-07 19:45 and 2018-08-08 14:33 EEST. You can find an excerpt from the Kamailio debug log at https://paste.fedoraproject.org/paste/bvyUMj3s5g2CcBN3JSBt5A/raw. As you can see, it crashes when trying to run the Lua route on an incoming packet. I am assuming the problem is this: Aug 8 14:33:14 lb-dev1 kamailio[13209]: *** Error in `/usr/sbin/kamailio': realloc(): invalid next size: 0x00005573dbff1ef0 *** However, I could not find any (direct) `realloc()` calls in the source for app_lua, so maybe this is an issue somehwere in liblua... Thoughts? Ivaylo

1 1

USRLOC db_mode=4 and server_id
by Darren Williams 13 Aug '18

13 Aug '18

I am currently using multiple instances of Kamailio with a common database and using the usrloc module with db_mode=4 and server_id to filter the contacts that get loaded at startup on each server. The problem I seem to have is that using the RPC call ul.add there I cannot see any parameter to send to specify the server_id. The server_id that the contact is given is 0. If I manually go into the db and modify the server_id everything works as expected. I think I must be missing something somehwere as I'mn sure you shouldn't need to do this. Can anyone help please. Thanks Daz

1 0

Kamailio TLS + letsencrypt
by Amar Tinawi 13 Aug '18

13 Aug '18

Hello i'm wondering if i can generate tls certificate for kamailio using letsencrypt with test domain name, in testing enviroment following this post : https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ got an error by letsecrypt because of checking of the domain name authority and no certificate was generated. Regards

3 2

mysql version
by Wilkins, Steve 12 Aug '18

12 Aug '18

Hello All, When Kamailio installs, it installs mysql 8.0. I would like it to install mysql 5.7. Is there a way to force this? Thank you

2 1

MYSQL and various
by Laura 12 Aug '18

12 Aug '18

Dear list, on my kamailio 4.4.7 i use interactions with MYSQL. For example .. i use it for some black list control and other like this.. sql_query("cb", "select id from blacklist where cli='$var(g2)' and (ippeer = '$si' or ippeer = 'ALL')", "rabl"); if($dbr(rabl=>rows)>0){ sql_result_free("rabl"); sl_send_reply("403","Forbidden blacklist"); exit; } sql_result_free("rabl"); I was wonder what was the best method to prevent the query going on if mysql don't respond for high load or other situation.. If Mysql is not able to reply to the query tight now the call continue over the configuration... what is the best way to prevent this and for examples drop it with a standard SIP 503 error. I have search on docs.. but i haven't found any valid way. Thanks Laura

2 1

problem with rewriteuri
by Eric Hosch 12 Aug '18

12 Aug '18

If I manually pass a string value to rewriteuri it works, but if I pass the same string from a database value it doesn't: if ($dbr(ra=>[0,4]) == "callfwd_noanswer") { $ does not work --> rewriteuri("$dbr(ra=>[0,6])"); works --> rewriteuri("sip:someone@somewhere"); append_branch(); route(0); Eric Hosch Chief Engineer KLBK-TV | KAMC-TV 7403 S. University Ave. Lubbock, TX 79423 O * 806.748.2208<tel:806.748.2208> F * 806.748.2269<tel:806.748.2269> ehosch(a)klbk13.tv<mailto:ehosch@klbk13.tv> www.everythinglubbock.com<http://www.everythinglubbock.com/> [Nexstar Media Group, Inc. | everythinglubbock.com | KLBK-TV | KAMC-TV] Nexstar Media Group, Inc. provides services for KAMC-TV and Mission Broadcasting, Inc. through an Outsourcing Agreement. KLBK and KAMC do not discriminate in advertising contracts on the basis of race, ethnicity or gender and further requires that in the performance of all KLBK/KAMC advertising agreements. KLBK/KAMC requires that each party not discriminate on the basis of race or ethnicity. This e-mail and any files transmitted with it are the property of NexstarMedia Group, Inc., are confidential, and are intended solely for the use of the individual or entity to whom this e-mail is addressed and/or as indicated in the applicable file. If you are not one of the named recipient(s) or otherwise have reason to believe that you have received this message in error, please notify the sender and delete this message immediately from your computer. Any other use, retention, dissemination, forwarding, printing or copying of this e-mail is strictly prohibited.

2 1

Nat issue
by eyas barhouk 11 Aug '18

11 Aug '18

hello dears when i'm using kamailio 5.2 as IMS i'm facing the following NAT issue : * when p-cscf handle a call from user1 to user2 that pre-registered on private address "192.X.X.X" (on another LAN) the user can't receive the invite message . in another word: proxy ip : 10.192.X.X user1 ip : 10.X.X.X user2 ip : 192.X.X.X the call scenario : user1 ---invite----> user2 (not arrived) * when p-cscf handle a call from user2 that pre-registered on private IP "192.X.X.X" (on another LAN) to user1 the user receive the invite message . the call scenario : user2 ---invite----> user1 (arrived) * when both of the user on different LANs the message 200 OK doesn't arrive to proxy :\ so based on that does any one has an experiment with NAT handling & nathelper module and how to configure it on P-cscf . thanks in advance

2 2

Ratio between cores ram and kamailio limits
by Renato Gallo 10 Aug '18

10 Aug '18

Hello, We're here to ask you what's the ratio between cpu cores/ghz, ram and how many contemporary calls kamailio can manage before it maxes out its capabilities and crash. Kind Regards Renato Gallo System Engineer sede legale e operativa: Via San brunone, 13 - 20156 - Milano (MI) Tel. +39 02 - 87049490 Fax +39 02 - 48677349 Mobile. +39 342 - 6350524 Wi | FreeNumbers: https://freenumbers.way-interactive.com Wi | SMS: https://sms.way-interactive.com Wi | Voip: https://voip.way-interactive.com Asterweb: http://www.asterweb.org Le informazioni contenute in questo messaggio e negli eventuali allegati sono riservate e per uso esclusivo del destinatario. Persone diverse dallo stesso non possono copiare o distribuire il messaggio a terzi. Chiunque riceva questo messaggio per errore è pregato di distruggerlo e di informare immediatamente [ mailto:info@sigmaware.it | info@ ] asterweb.org

2 1

UDP workers block when one or more rtpengine instances go offline
by Sebastian Damm 10 Aug '18

10 Aug '18

Hi, we run multiple rtpengine servers to share the load. Whenever we need to take an rtpengine server offline, we used to just block the control port via iptables, then no new calls ended up on this instance of rtpengine. This worked pretty well in Kamailio 4.4.5. However, since Kamailio 5.0, and the problem persists with 5.1.4, Kamailio hangs almost immediately after we block the control port traffic. In the log file there are almost no packets processed except every few seconds, which looks like a timeout thing. Did we configure anything wrong there? Or is the "dead rtpengine detection" just broken? Our configuration: loadmodule "rtpengine.so" modparam("rtpengine", "rtpengine_disable_tout", 120) modparam("rtpengine", "setid_avp", "$avp(rtpsetid)") modparam("rtpengine", "rtpengine_sock", "0 == udp:1.2.3.4:9001=2 udp:1.2.3.5:9001=2 udp:1.2.3.6:9001=2") modparam("rtpengine", "rtpengine_sock", "1 == udp:2.3.4.5:9001=2") Any help appreciated. Regards, Sebastian

4 11

kamcmd htable.reload propertiesHash problem
by Ginhoux, Patrick 10 Aug '18

10 Aug '18

Hi, I run Kamailio 5.0.1 with MariaDB database. There is no issue at the initialization, meaning the db is opened by kamailio and the hash tables loaded from the db tables. Example: # kamcmd htable.dump propertiesHash { entry: 1 size: 1 slot: { item: { name: cfg.env But the htable.reload command fails: error: 500 - Failed to open htable db connection The Kamailio log reports that it can't connect the db : DEBUG: ctl [../../core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0x7f68c74ab300, 10, 3, 0xdc9410), fd_no=1 DEBUG: ctl [io_listener.c:442]: handle_new_connect(): handle_stream read: new connection (1) on /var/run/kamailio//kamailio_ctl DEBUG: <core> [core/sr_module.c:694]: find_mod_export_record(): find_export_record: found <db_bind_api> in module db_mysql [/usr/lib64/kamailio/modules/db_mysql.so] DEBUG: <core> [db.c:209]: db_bind_mod(): using db bind api for db_mysql DEBUG: <core> [db.c:314]: db_do_init2(): connection 0x7f68cc354c10 not found in pool DEBUG: db_mysql [km_my_con.c:98]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@/siprouter ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2 "No such file or directory") ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool ERROR: htable [ht_db.c:92]: ht_db_open_con(): failed to connect to the database My local db is located at /myqdata/mysiprdb/data where is the mysiprdb.sock file. But the log shows that Kamailio is looking for /var/lib/mysql/mysql.sock. Does this error make sense to you ? Is it possible to fix this problem by configuration in Kamailio ? My kamailio.cfg script contains : # ----- htable modparam("htable", "db_url", "mysql://[kamailio]/siprouter") modparam("htable", "fetch_rows", 1000) modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") In the /myqdata/mysiprdb/data/my.cnf, I added the following entries for Kamailio: [kamailio] socket = /myqdata/mysiprdb/data/mysiprdb.sock user = siprouter password = ########## default-character-set = utf8 Thanks all for help. Cordialement. Patrick GINHOUX

2 12

Kamailio troubleshooting
by Sergey Safarov 09 Aug '18

09 Aug '18

One my server with perl external script time to time is stopping processing SIP messages. Perl script make calls to kamailio database on mysql. I have found this in logs DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0x56427eb40c40, 495, 2, 0x7f1a9f9ba408), fd_no=33 DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0x56427eb40c40, 495, -1, 0x0) fd_no=34 called DEBUG: <core> [core/tcp_main.c:4196]: handle_tcpconn_ev(): sending to child, events 1 DEBUG: <core> [core/tcp_main.c:3871]: send2child(): WARNING: no free tcp receiver, connection passed to the least busy one (56) DEBUG: <core> [core/tcp_main.c:3875]: send2child(): selected tcp worker 4 20(24) for activity on [tcp:0.0.0.0:81], 0x7f1a9f9ba408 And no next messages. How get info what is load on "tcp receiver". This host in dev mode and handles not more 5 devices. Sergey

2 2

Re: [SR-Users] problem with compiling kamailio version 5.1.4 including a private Module
by Abdulaziz Alghosh 09 Aug '18

09 Aug '18

Hello Henning, thanks a lot for your response and explanation. strutils.[c,h] and strcommon.[c,h] are almost the same. So I modified my included header from stcommon[] to strutils as you mentioned. Unfortunately, I surfed sequentially Kamailio version upgrades. Even though, i did not found such hints about this modification. So I have some doubts the called headers and codes from our own module. Hence, I am asking for yor advice. What I am trying to do now is compiliing our module separately under it is own directory so I gain the Shared Object file (.so). Somehow our Module is requring cds, presence and xcap libraries. But the strange thing is that headers from these libraries are #including other headers assuming these included header files are under the same library's directory (but in fact most of the header files are under ../src/core/). For example : Making in cds CC (gcc) [L libser_cds.so.0.1] cds.o In file included from cds.c:2: /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:21: warning: mem/mem.h: No such file or directory /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:25: warning: mem/shm_mem.h: No such file or directory In file included from cds.c:3: /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: warning: locking.h: No such file or directory In file included from cds.c:4: /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20: error: dprint.h: No such file or directory make[2]: *** [cds.o] Error 1 make[1]: [cds] Error 2 (ignored) Those libraries are delivered with version 5.1.4 itself so i do not think it is a good idea to modify the paths of the included header files to compile my module. because most probably these libraries are needed by standard default modules of kamailio. I hope this is not too much for you to inspect. Note: The attached has the log files ( resulting from make all and install ) . Your kind response is highly appreciated BR Abdulaziz On Thu, Aug 9, 2018 at 11:31 AM, Abdulaziz Alghosh <aziz647(a)gmail.com> wrote: > Hello Henning, > > thanks a lot for your response and explanation. strutils.[c,h] and > strcommon.[c,h] are almost the same. So I modified my included header from > stcommon[] to strutils as you mentioned. > Unfortunately, I surfed sequentially Kamailio version upgrades. Even > though, i did not found such hints about this modification. So I have some > doubts the called headers and codes from our own module. Hence, I am asking > for yor advice. > > What I am trying to do now is compiliing our module separately under it is > own directory so I gain the Shared Object file (.so). Somehow our Module is > requring cds, presence and xcap libraries. But the strange thing is that > headers from these libraries are #including other headers assuming these > included header files are under the same library's directory (but in fact > most of the header files are under ../src/core/). For example : > > Making in cds > CC (gcc) [L libser_cds.so.0.1] cds.o > In file included from cds.c:2: > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:21: > warning: mem/mem.h: No such file or directory > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:25: > warning: mem/shm_mem.h: No such file or directory > In file included from cds.c:3: > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: > warning: locking.h: No such file or directory > In file included from cds.c:4: > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20: > error: dprint.h: No such file or directory > make[2]: *** [cds.o] Error 1 > make[1]: [cds] Error 2 (ignored) > > > Those libraries are delivered with version 5.1.4 itself so i do not think > it is a good idea to modify the paths of the included header files to > compile my module. because most probably these libraries are needed by > standard default modules of kamailio. > > I hope this is not too much for you to inspect. > Note: The attached has the log files ( resulting from make all and install > ) . > > Your kind response is highly appreciated > Abdulaziz > > > On Thu, Aug 9, 2018 at 8:11 AM, Henning Westerholt <hw(a)kamailio.org> > wrote: > >> Am Mittwoch, 8. August 2018, 17:47:07 CEST schrieb Abdulaziz Alghosh: >> > i am somehow newbie with Kamailio and trying to migrate from kamailio >> > version 3.0.3 to 5.1.4. Former fellows had developed own modules which >> > needed "strcommon.h" and "strcommon.c". >> > >> > These last two files are not delievered with version 5.1.4. but I copied >> > them under ../src/core/ thinking it would be beneficial. Unfortunately, >> > after make prefix / (and including our own module) all, it seem that >> > several dependencies exist. Especially from header files under >> > ..src/lib/xcap/ and ../src/lib/presence >> > >> > Can somebody tell me how "strcommon.h" and "strcommon.c" where >> replaced in >> > version 5.1.4 ? >> >> Hello Abdul, >> >> the functions in strcommon.[c,h] were moved into src/core/strutils.[c,h] >> in >> releases starting from 5.0. I'd suggest to have a look to the file to >> verify >> if there are no other changes. >> >> Best regards, >> >> Henning >> -- >> Henning Westerholt >> https://skalatan.de/blog/ >> > >

1 0

problem with compiling kamailio version 5.1.4 including a private Module
by Abdulaziz Alghosh 09 Aug '18

09 Aug '18

Greetings, i am somehow newbie with Kamailio and trying to migrate from kamailio version 3.0.3 to 5.1.4. Former fellows had developed own modules which needed "strcommon.h" and "strcommon.c". These last two files are not delievered with version 5.1.4. but I copied them under ../src/core/ thinking it would be beneficial. Unfortunately, after make prefix / (and including our own module) all, it seem that several dependencies exist. Especially from header files under ..src/lib/xcap/ and ../src/lib/presence Can somebody tell me how "strcommon.h" and "strcommon.c" where replaced in version 5.1.4 ? Best regards Abdul Aziz

2 1

Questions about changes between 5.0.1 and 5.0.7
by Ginhoux, Patrick 08 Aug '18

08 Aug '18

Hi, Following a problem with the kamcmd htable.reload with Kamailio 5.0.1 (see related thread for this), I have upgraded to the 5.0.7 level (with some difficulties due to my specific environment). My server runs the RHEL 7.4 distribution. Now the upgrade is complete, I notice that that are changes not really documented: - The kamailio-perl-5.0.7-1.el7.x86_64.rpm requires to install the mod_perl module that doesn't exist anymore in RHEL 7. It needs to be installed from the EPEL repository - The init script for kamailio has been changed from the old method to the fully compliant method for system unit for RHEL 7 for example: With the 5.0.1 level we have the "/etc/rc.d/init.d/kamailio" With the 5.0.7 level we have now the "/usr/lib/systemd/system/kamailio.service" So my questions/comments are : - Why the mod_perl module is required between 5.0.1 and 5.0.7 (question posted in the attached message) ? - In the previous kamailo init script, I added a condition to start MySQL before kamailio by adding this entry in the script : # Required-Start: $syslog $network $mysqld The new script contains the After command : [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=network.target Should I change the After line with this : After=network.target mysql.target to have the same condition? In all the cases, some documentation must be changed to provide more details. Thanks in advance for the answers. Cordialement. Patrick GINHOUX

3 16

RTPEngine UDP receive queue issue
by Alex Balashov 08 Aug '18

08 Aug '18

Hi, We have an RTPEngine that occasionally finds itself in this state of CPU usage: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 25476 root 20 0 2929600 80820 1900 S 156.2 0.1 259:05.54 rtpengine The journal is a dizzying stream of: Aug 06 20:58:47 gw1 rtpengine[25476]: ERR: [1074537188_44545952 port 61666]: Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible It eventually leads to increasing CPU usage and a degeneration of audio quality until RTPEngine is restarted. It would be appreciated if someone could shed some light on the nature of this queue and what we can do about this problem. Is this referring to a netfilter packet forwarding queue, or is it a receive queue of the RTPengine user space control process? Can its size be increased or tweaked with kernel parameters? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 10

sipwise/sems for video
by Amar Tinawi 08 Aug '18

08 Aug '18

Hello i don't know if the question is related to this list, but as far as i'm using Kamailio i though i should ask here i'm using Kamailio and sems as media server, the audio is passing through fine but when it comes to Video, SEMS block it, in the outgoing SDP m=video is removed with it's codec my question is does sems support Video ? or is there any tuning for sems to allow Video media streams ? Cheers Amar

2 3

Regarding OpenSource Meetup in India
by surendra tiwari 08 Aug '18

08 Aug '18

Hello Guys, We are having a Meetup for Telecommunication Geeks. We are talking around Opensource Technology for building telecommunication solution. Here's the agenda for our first meetup. We do have one slot open. Please contact us or drop a comment if you want to present. Hoping that everyone would have fun geeking out on telecom and voip tech. 10:00 - 10:25 - How VOIP is changing the Communication landscape - Rajat Dwivedi 10:30 - 10:55 - Generic VoIP Solutions on the cloud - Chandra Prakash Singh 10:55 - 11:10 - Break 11:10 - 11:35 - Developing VoIP solutions using Open Source Tech - Surendra Tiwari 11:40 - 12:05 - WebRTC: Telephony in your browser - Altanai Bisht 12:05 - 12:30 - Open Slot (looking for speakers) 12:30 - 1:30 - Lunch + Networking 13:30 - 14:15 - Birds of a Feather Session (BoF) Please RSVP. With Regards, Surendra Tiwari VoIP Application Developer, Plivo Comm +91-9967609476

3 2

MSILO messages to users behind NAT
by Michalis Michael 07 Aug '18

07 Aug '18

I am using Kamailio (5.0.1) and MSILO module for offline messages but I have a problem with users behind NAT. According to module overview, MSILO is forwarding the offline messages to contact address provided in REGISTER request. Is there any way to overcome this and send the messages to the external client IP? Thanks, Michalis Michael

1 0

IRC Channel
by Enrico Bandiera 07 Aug '18

07 Aug '18

Hi, currenty (the last few days) there has been a spam attack on freenode, #Kamailio is currently full of spam messages, I was wondering if who has operator rights could set the channel to +r (only registered users can send messages) as suggested by freeenode admins The channel usually has only few messages and with all this spam it becomes impossible at all to use it. Greetings, Enrico.

4 4

Kamctl Stats Update
by Noah Mehl 07 Aug '18

07 Aug '18

Hey all, I was wondering if the kamctl stats output could be updated to provide a bit more parseable JSON? For instance: # kamctl stats shmem | jq . { "jsonrpc": "2.0", "result": [ "shmem:fragments = 18", "shmem:free_size = 467187808", "shmem:max_used_size = 69694104", "shmem:real_used_size = 69683104", "shmem:total_size = 536870912", "shmem:used_size = 41048984" ], "id": 4300 } I would hope would be more like this: { "jsonrpc":"2.0", "result":[ { "shmem":[ { "fragments":18, "free_size":467187808, "max_used_size":69694104, "real_used_size":69683104, "total_size":536870912, "used_size":41048984 } ] } ], "id":4300 } I apologize in advanced because I don’t have the skillset to contribute such a feature. Thanks! ~Noah

5 8

"kamdbctl create" failing with MySQL 8
by Wilkins, Steve 05 Aug '18

05 Aug '18

Hello All, There appears to be issues with running "kamdbctl create" when using MySQL 8. When this is ran there are syntax SQL errors. One such error is when doing grants using "IDENTIFIED BY 'password'"; This throws a sequel error for version 8 of MySQL. Thank you,

2 1

DMQ problems
by Aleksandar Sosic 05 Aug '18

05 Aug '18

Hi, I've set up a dmq server for other kamailio nodes with this parameters: ``` modparam("dmq", "server_address","sip:10.0.0.101:5060") modparam("dmq", "notification_address", "sip:127.0.0.1:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 15) ``` On the kamailio nodes the config is the following: ``` modparam("dmq", "server_address", "sip:10.0.0.x:5060") modparam("dmq", "notification_address", "sip:10.0.0.101:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 15) ``` On the kamailio dmq server I've got this in the logs: ``` 25(41) ERROR: dmq [notification_peer.c:596]: notification_resp_callback_f(): deleting server sip:127.0.0.1:5060 because of failed request 25(41) ERROR: dmq [notification_peer.c:599]: notification_resp_callback_f(): not deleting notification_peer ``` On the kamailio nodes the same thing with the ip of the dmq server: ``` 25(112) ERROR: dmq [notification_peer.c:596]: notification_resp_callback_f(): deleting server sip:10.0.0.101:5060 because of failed request 25(112) ERROR: dmq [notification_peer.c:599]: notification_resp_callback_f(): not deleting notification_peer ``` The request route on the dmq server is simple as this: ``` request_route { if(is_method("KDMQ")){ dmq_handle_message(); } }#end request_route ``` I've tried with some xlog-s inside the if(is_method("KDMQ")) but they don't show up so the dmq_handle_message() is not even executed. I've also tried to bind the dmq server to a different port but with no success on making it work. The dmq.list_nodes shows only the dmq server IP and the current node IP, the IPs of other nodes are not shown: ``` root@proxy-94b6ccf46-t8fx2:/# kamcmd dmq.list_nodes { host: 10.0.0.101 port: 5060 resolved_ip: 10.0.0.101 status: active last_notification: 0 local: 0 } { host: 10.0.0.45 port: 5060 resolved_ip: 10.0.0.45 status: active last_notification: 0 local: 1 } ``` And it's the same on all other kamailio nodes: dmq server ip and local ip. The kamailio version on all the nodes is: 5.2.0~dev6+0~20180616010152.1138+xenial Thanks, -- Aleksandar Sosic mail: alex.sosic(a)evosip.cloud

4 20

Security announcement related to Kamailio
by Henning Westerholt 04 Aug '18

04 Aug '18

Hello, I want to highlight that the last stable versions (for the two maintained series: 5.0 and 5.1) include fixes for an security issues that can crash a running instance of Kamailio, therefore it is strongly recommended to upgrade. Details: There exists a security vulnerability in the Kamailio SIP server related to "To" header processing. A specially crafted SIP message with double "To" header and an empty "To" tag causes a segmentation fault and crashes Kamailio. The reason is missing input validation in the "build_res_buf_from_sip_req" core function. If an attacker sends many of this messages this would lead to a Denial of Service of the attacked infrastructure. This is especially critical as no authentication for the remote source is needed. You find all the details including a proof of concept code in the published security announcement in my blog: https://skalatan.de/blog/advisory-hw-2018-05 A CVE report for this issue is going to be created as well in the near future. The issues were found some months ago and were fixed quickly. The code related to the reported issues is rather old and there are no known incidents of exploiting these issues so far. However, once the CVE report becomes public, there could be a higher risk of exploitation. Please address any detailed technical questions related to this to the developer list at sr-dev(a)lists.kamailio.org . In case of confidential remarks related to this or other security issues, please address them to the Kamailio Management. Best regards, Henning Westerholt -- Henning Westerholt https://skalatan.de/blog/

1 1

Nathelper not obeying Advertised IP
by John Martirosian 03 Aug '18

03 Aug '18

Hello, The nathelper module does not obey the advertise IP. Is there any work around for this? When kamailio is running behind a NAT with advertised IP set the nathelper module doesn't use that IP when sending SIP OPTIONS keepalives. Here is a picture of the packet coming from Kamailio. https://i.imgur.com/9vcpOoo.png Thanks,

1 0

Final reply code is wrong in case of 302 and 486 busy
by Marco Capetta 03 Aug '18

03 Aug '18

Hello, we have a problem of final reply code in case of 302 and then 486 busy. This is our scenario: A, B and C are subscribers. A ----------> Kamailio ----------> B Kamailio <---302---- B Kamailio -----------------------> C Kamailio <---486----------------- C A <---302---- Kamailio Expected behavior: Kamailio should reply to A with 486, as C was busy, and not 302. We have kamailio v5.1.4 In the call to B we set both failure and branch_failure: t_on_failure("failure_route"); t_on_branch_failure("failure_redirect"); in order to handle the 302 redirect per branch with: event_route[tm:branch-failure:failure_redirect] { if($T_reply_code == 301 || $T_reply_code == 302) { do redirect ... } } and the final failure with: failure_route[FAILURE_ROUTE_LOCAL] { do something ... } Since we have 'modparam("tm", "failure_reply_mode", 3)' my expectation is to get the winning reply from the branch of last step. Is there any configuration I'm missing? Since we manage the redirect with a branch-failure, is the winning reply calculated in a different way? Thanks Marco

2 1

Other issues encountered with MySQL 8 and Kamailio
by Wilkins, Steve 03 Aug '18

03 Aug '18

Good Morning all, When I do get Kamailio to compile with MySQL 8, I encounter the following issue when running "kamdbctl create" ERROR 1449 (HY000) at line 1: The user specified as a definer ('mysql.infoschema'@'localhost') does not exist WARNING: Your current default mysql characters set cannot be used to create DB. Please choice another one from the following list: Any ideas? Thank you.

1 1

5.1 compilation error on Centos 7
by Wilkins, Steve 03 Aug '18

03 Aug '18

Hello all, I started getting the following error while trying to compile Kamailio 5.1+ on Centos 7 CC (gcc) [M db_mysql.so] my_fld.o In file included from my_fld.c:22:0: my_fld.h:37:2: error: unknown type name 'my_bool' my_bool is_null; Has anyone seen this before. Thank you

2 2

is_method not matching message type KDMQ breaking DMQ bus. (master branch)
by Asgaroth 02 Aug '18

02 Aug '18

Hi All, I've come across and oddity whereby my is_method("KDMQ") checks are not matching. I'm testing with kamailio master: version: kamailio 5.2.0-dev6 (x86_64/linux) a16272 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: a16272 compiled on 15:27:21 Jul 31 2018 with gcc 7.3.1 Sample routing logic: request_route { route("InitialMessageChecks"); exit; } route[InitialMessageChecks] { xlog("L_INFO", "Received message type $rm from $si:$sp on $Ri:$Rp"); if ( is_method("KDMQ") ) { xlog("L_INFO", "KDMQ check passed."); } if ( is_method("OPTIONS") ) { xlog("L_INFO", "OPTIONS check passed."); } if ( is_method("SUBSCRIBE") ) { xlog("L_INFO", "SUBSCRIBE check passed."); } } In the logs I am only receiving the 1st log line in the initial message checks route for message types of KDMQ, eg: INFO: <script>: Received message KDMQ from 1.1.1.1:5060 on 1.1.1.2:5060 INFO: <script>: Received message KDMQ from 1.1.1.3:5060 on 1.1.1.2:5060 I never seem to hit the is_method check log line when the method type is KDMQ. I do see the options/subscribe logs though: INFO: <script>: OPTIONS check passed. INFO: <script>: SUBSCRIBE check passed. nothing for KDMQ. Reason for asking is: this check never matches and breaks the dmq bus: if ( is_method("KDMQ") ) { if ( dmq_is_from_node() ) { dmq_handle_message(); exit; } else { exit; } } Any thoughts, or am I going crazy :) Thanks

3 4

Re: [SR-Users] Method Destination Variable
by Duarte Rocha 02 Aug '18

02 Aug '18

Greetings, I thought about that but i wasn't sure it would be safe. But, in this case, the Request-URI domain will be the only place where Kamailio will look to find out the destination right within the dialog right? I'm going the statelless way, so that might be the correct answer. Thanks for your help

2 2

get_out_socket(): no socket found
by Ilie Soltanici 02 Aug '18

02 Aug '18

Daniel, Not, those IP are in a different subnet. There is a firewall - running on a different device. Ping and SIP Messages are going very well from 2.2.2.2 to 8.8.8.8. I was able to solve this issue by disabling mhomed and set "listen" external interface with IP address 2.2.2.2 on top in the configuration file. Now, Kamailio is starting without any errors and is registering to the upstream provider, but I just I am wondering if it's possible to specify an outgoing socket for uac module in the configuration file? Thank you.

1 0

get_out_socket(): no socket found
by Ilie Soltanici 02 Aug '18

02 Aug '18

Daniel, Yes, IP Address has been edited, 2.2.2.2 and 8.8.8.8 are not real IP Address-es, this is just to show that those IP are external IP's. Regards, Ilie

2 1

Method Destination Variable
by Duarte Rocha 01 Aug '18

01 Aug '18

Greetings, I have a Kamailio server set up without Registers. I'm using Dispatcher module for balancing and selecting the destinations. I need to do some changes in the message according to the chosen destination. If it's the first INVITE, after Dispatcher operations, i can do it using $dd or $du variables. However, in later messages, like a BYE, i can't seem to access the method's destination IP. Is there any variable that i'm missing? I've tried $route_uri but it only returns the kamailio's own IP, since it's the only IP in the "Route" header. Thanks in advance, Cheers

2 1

get_out_socket(): no socket found
by Ilie Soltanici 01 Aug '18

01 Aug '18

Hi All, I'm unable to register an extension to the Upstream Provider by using uac module. Getting the following error: *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: <core> [core/forward.c:181]: get_out_socket(): no socket found* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: <core> [core/forward.c:183]: get_out_socket(): no corresponding socket found for(udp:8.8.8.8:5060 <http://8.8.8.8:5060>)* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: tm [ut.h:317]: uri2dst2(): no corresponding socket found for "svr04.nuacom.net <http://svr04.nuacom.net>" af 2 (udp:8.8.8.8:5060 <http://8.8.8.8:5060>)* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: tm [uac.c:449]: t_uac_prepare(): no socket found* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: uac [uac_reg.c:1161]: uac_reg_update(): failed to send request for [1050]* Kamailio is listening on 2 interfaces: [root@kamailio user]# /usr/local/sbin/kamcmd core.sockets_list { socket: { proto: udp address: 10.1.1.11 ipaddress: 10.1.1.11 port: 5060 mcast: no mhomed: no } socket: { proto: udp address: 2.2.2.2 ipaddress: 2.2.2.2 port: 5060 mcast: no mhomed: no } mhomed is enabled in the configuration file: *mhomed=1* I'm not using ' force_send_socket' anywhere in the configuration file. This is the uac module configuration: modparam("uac", "reg_keep_callid", 1) modparam("uac", "reg_contact_addr", "2.2.2.2") modparam("uac", "reg_db_url","cluster://kamailio") If i'm disabling mhomed from configuration file - i don't receive any errors - but then kamailio is trying to send the message to the external server (8.8.8.8) through internal interface (10.1.1.1). Thank You.

2 1

kamalio in virtual environment
by AL S 31 Jul '18

31 Jul '18

Hi, Is recent kamalio release interoperable in virtual environment? Thanks, Alex

5 5

How to get Formatted Current TIME (NOT CACHED)
by Alvaro Araos 31 Jul '18

31 Jul '18

Hi, If i call to $timef multiple times, always returns the same time string (CACHED), how i get formatted current time?? Example: $dlg_var(start)=$timef(%Y-%m-%d %H:%M:%S) // RETURNS "2018-07-28 18:44:35" .. some seconds later $dlg_var(start)=$timef(%Y-%m-%d %H:%M:%S) // RETURNS THE SAME DATE STRING "2018-07-28 18:44:35" Saludos, ALVARO ARAOS MÉNDEZ Director de Tecnología en FONOIP +56 2 2947 9000 alvaro(a)fonoip.com | www.fonoip.com [1] Moisés Chacón 46, El Monte, RM Links: ------ [1] http://www.fonoip.com

1 1

RTP Proxy
by Alex Tatham 31 Jul '18

31 Jul '18

Hello everyone, I'm in the process of moving approx. 100 Kamailio/Media Proxy (AG Projects) servers to Kamailio/RTPEngine. I have all the functionality in place and I have a handful of servers in production. I'm trying to monitor the RTP engine (using Munin) and I'm graphing the values returned from the RPC command rtpengine.get_hash_total. This value is slowly increasing over time and I'm trying to find out if this is something I should worry about. I want to make sure that sessions are not being left open and therefore the software will eventually run out of ports. To give some values, if I make 100 calls the call will be around 25 after those calls. >From all other tests I have done including a check through the verbose logs I can see the sessions are being closed at the end of the calls. Can anyone help with this at all or does anyone possibly have any better ideas? Thank you, Alex Alex Tatham Technical Director T:01233 220 943 E:alex.tatham@dmcplc.co.uk W:www.dmctechnologies.co.uk DMC Technologies formerly HP Technology Unit 5, Invicta Business Centre | Orbital Park | Ashford | Kent | TN24 0HB DMC Technologies is a trading name of Hobbs Parker 2000 Limited Registered Office: Romney House, Orbital Park, Monument Way, Ashford TN24 0HB Registered in England No.04027755

2 1

Is anyone having issues with dialog metrics when DMQ is enabled?
by Joel Serrano 30 Jul '18

30 Jul '18

Hi guys, Is anyone having issues gathering active/early dialog metrics from kamailio when DMQ is enabled? I have an open ticket in GH ( https://github.com/kamailio/kamailio/issues/1591) and I'm wondering if this could be happening to anyone else? So far the theory is that the counter goes below 0 and that creates the problem. I'd love to hear from you if you are experiencing this too. If any of you are using Kamailio 5.1 + DMQ and gather active/early dialog metrics using the stats.fetch or stats.get_statistics rpc commands, do you ever get values close to 64BIT ULONG_MAX? (18446744073709551615)?? Thanks! Joel.

2 3

Kamailio 5.0.X RPM packages/repo broken
by Patrick Wakano 27 Jul '18

27 Jul '18

Hello list, Hope you all doing fine! I've been trying to install Kamailio packages from the opensuse repo (as per https://www.kamailio.org/wiki/packages/rpms) but the installation is failing with this message: Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-ims-5.0.7-1.el7.cento FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/C…: [Errno 14] HTTP Error 416 - Requested Range Not Satisfiable --:--:-- ETA Trying other mirror. kamailio-5.0.7-1.el7.centos.x8 FAILED 77% [============================================================- ] 1.0 MB/s | 4.9 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.0.x-rpms clean metadata Trying other mirror. I've successfully installed the 5.0.7 version on Friday (06/07) but I've noticed the packages have changed last Saturday (07/07/2018).... Given this error message looks like the repository is broken somehow.... Is anyone aware of such issue? I haven't tried the 5.1.X repo because my script was not migrated to 5.1.X yet.... Thanks, Kind regards, Patrick Wakano

3 13

NGN and NNI functionalities
by Duarte Rocha 27 Jul '18

27 Jul '18

Greetings, Does Kamailio has any module that deals with NGN and NNI ? Also, is there any variable or transformation that gives me access to a parameter in the User Part of an URI ? Best Regards, Duarte Rocha

2 1

No cdr after Kamailio 5.0.4 and 5.2 dev restart
by Kordován Szabolcs 27 Jul '18

27 Jul '18

Hi All, In March there was a discus about this issue. We tested the new 5.2 dev and 5.0.4 version of Kamailio and we experience that If we restart Kamailio during the call the cdr won't be stored in table acc_cdrs but the data will be inserted to table acc and data will be deleted from table dialog and dialog_vars. Did I omit the fix for it? Thanks for your help. Best regards, Szabolcs

2 3

Package hashes for Kamailio 5.1.4 rpms are wrong.
by Skip Morse 26 Jul '18

26 Jul '18

Hi, The package hashes for Kamailio 5.1.4 rpms are wrong. I actually saw this happen with 5.1.3 and then somehow I think it got corrected. Repo details: http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C… Filename: kamailio-5.1.4-2.el7.centos.x86_64.rpm Path: /repositories/home:/kamailio:/v5.1.x-rpms/CentOS_7/x86_64/kamailio-5.1.4-2.el7.centos.x86_64.rpm Size: 5.4M (5645948 bytes) Last modified: Thu, 19 Jul 2018 02:53:15 GMT (Unix time: 1531968795) SHA-256 Hash: aed812fa3ccf502634fe6de848eca2ce167ee70f6ace31aa1541a5787502667d SHA-1 Hash: ae0977ca32e6b5ba2a627e58948098f386639582 MD5 Hash: 68784949f33d6259420b50de5e70a4ec BitTorrent Information Hash: f743a7c45d50065084d40a314bd80e3ec143c384 What I get locally: $ wget http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… $ sha256sum kamailio-5.1.4-2.el7.centos.x86_64.rpm a21d2b6c98da041af85ebc0f8b6fe8525e8e247e84317c32db021e90ec43b209 kamailio-5.1.4-2.el7.centos.x86_64.rpm $ sha1sum kamailio-5.1.4-2.el7.centos.x86_64.rpm 9cbf6de153fb8ffc4913f0582d6e85535b44975f kamailio-5.1.4-2.el7.centos.x86_64.rpm $ md5sum kamailio-5.1.4-2.el7.centos.x86_64.rpm d36149f4bd91d0879c239b8e81264fb3 kamailio-5.1.4-2.el7.centos.x86_64.rpm Let me know if I should report this elsewhere. Thanks! -Skip

1 0

DMQ with dialog: KDMQ messages being looped back to the source
by Patrick Murphy 26 Jul '18

26 Jul '18

Hi, I am using Kamailio 5.1.1 with DMQ with the following config (relevant parts) on 2 nodes. Node A: modparam("dialog", "enable_dmq", 1) modparam("dmq", "server_address", "sip:no.de.a.ip:5060") modparam("dmq", "notification_address", "sip:myvoip.local") modparam("dmq", "multi_notify", 1) Node B: modparam("dialog", "enable_dmq", 1) modparam("dmq", "server_address", "sip:no.de.b.ip:5060") modparam("dmq", "notification_address", "sip:myvoip.local") modparam("dmq", "multi_notify", 1) "myvoip.local" is a DNS A record that resolves to both the IPs of A and B. I have the dmq_handle_message(); called if method is of type "KDMQ" in the request_route. Now, in order to test it, I only sent a few calls only on node A. KDMQ messages flowed to node B and dialog state replicated. But I noticed, node B was randomly sending one of the KDMQ dialog messages back to A and receiving a 500 "Server Internal Error" back from node A... (for a subset of those calls, not all). Why would this be the case when the nodes correctly recognize their local status by the "local: 1" set as seen from dmq.list_nodes Anyone else experienced this? What could I be doing wrong? Thanks, Pat.

2 5

FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides
by Giovanni Maruzzelli 25 Jul '18

25 Jul '18

Hello fellow RTCers, just finished presenting about scaling videoconferencing, chat and moderation/direction controls in SIP, with web clients, deskphones, smartphone apps, server push (google fcm) via cordova and SIP.js at www.cluecon.com. Here's the slides of my presentation: http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_… or, shorter, http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf >From ClueCon blurb: FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service On top of traditional SIP deskphones and softphones, WebRTC let us extend the reach of SIP to all browsers, and to smartphone apps that wake up with a server push. OpenSIPS has superb SIP and WebRTC support and can protect, balance and scale FreeSWITCH insuperable conferencing, video MCU, media switching/mixing, SIP SIMPLE message processing capabilities. We will see how to build and implement such a complete platform, touching both on servers and on clients. Hint: SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to transport statuses, commands and chats between participants and administrators of SIP audio/video calls/conferences. Giovanni MaruzzelliPrincipal at OpenTelecom.IT ------------------------------ Giovanni is a consultant for the Telco sector, developing software and training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is heavily engaged with FreeSWITCH, of which he wrote the interface with Skype and with cellular phones. An Internet tech pioneer, in 1996 Giovanni was cofounder of Italia Online, the most popular Italian portal and consumer ISP, and architect of its Internet technologies – www.italiaonline.it Then supervisor of Internet operations and architect of the first engine for paid access to www.ilsole24ore.com , the most read financial newspaper in Italy and to its databases (migrated from mainframe). After that, he was CEO of venture capital funded Matrice, developing Telemail unified messaging and multi language phone access to email (Text To Speech), and CTO of incubator funded Open4, an open source managed applications provider. Then he was for two years in Serbia as Internet and Telecommunication Investment Expert for World Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco companies worldwide. -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

3 4

sipdump excluding dmq exchanges
by Patrick Murphy 24 Jul '18

24 Jul '18

Hi Can we have sipdump only log SIP messages and not KDMQ? Thanks, Pat.

2 1

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