sr-users September 2018

sr-users@lists.kamailio.org

72 participants
127 discussions

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

2 2

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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sr-users September 2018