sr-users September 2018

sr-users@lists.kamailio.org

72 participants
122 discussions

by Gertjan Wolzak

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

5 years, 7 months

Cannot handle Cancel Request properly

by Aqs Younas

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.

5 years, 7 months

replay of tls delayed by 30 secs.

by Andrew Chen

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.*

5 years, 7 months

subst_hf to rewrite a contact header

by Ben Hood

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

5 years, 7 months

Documentation of Kamailio.cfg

by Prabhat Kumar

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…>

5 years, 7 months

Kamailio - Asterisk integration

by denislav.tsonev＠musala.com

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

5 years, 7 months

Problem with replicating dialogs using DMQ

by jenus＠cyberchaos.nl

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

5 years, 7 months

create domain "on the fly" in registrar modukle

by Maarten Ureel

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

5 years, 7 months

Adding new rtpproxy instance without restarting Kamailio

by Pintu Lohar

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

5 years, 7 months

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

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

5 years, 7 months

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sr-users September 2018