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42 participants
37517 discussions

Kamailio RPC configuration
by sadik.oualla.mohamed＠gmail.com 24 Feb '23

24 Feb '23

Dear Kamailio community, I am a newbie in Kamailio and I am currently working on a project that involves communicating with Kamailio using Python. I have been exploring the Kamailio documentation and I have come across RPC, JSONRPCS..., which I believe can help me achieve my goal. However, I am not sure about the steps to follow in the Kamailio configuration file to create my own script to communicate with Kamailio using Python (the script is on an other machine on the same LAN). I would be grateful if someone could guide me through the process. I have already set up Kamailio (builed it from the source code 5.5.5) and Python on my machines and I have basic knowledge of Kamailio configuration. I just need some guidance on how to set up the RPC module in Kamailio and how to create a Python script to communicate with it. I found that I can use RPyC library of python. I would appreciate any help or suggestions from the Kamailio community. Thank you in advance for your time and assistance. Best regards, Wild Coder.

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OT: MS Teams source IP address ranges
by Jon Bonilla (Manwe) 24 Feb '23

24 Feb '23

Hi Sorry for the OT but I think here's the place where I an find a lot of Ms teams integrations I've been working on MS teams direct routing integration for PekePBX. It works. I guess I've done it as everybody else, using Henning's guide as base and extending it for multitenant setup (thanks Henning!) What I've realized is that the source IP address of calls coming from MS are not always matching dispatcher hosts. Sometimes they come from another source IP and failover to the dispatcher hosts when they receive no response. That makes some of the calls to have an additional latency Searching in the MS doc I see that they document these nets as source of their signaling: 52.112.0.0/14 52.120.0.0/14 But I've seen IP addresses outside of this range as source. In this blog https://erwinbierens.com/microsoft-teams-direct-routing-ip-addresses/ The ranges are listed as 52.112.0.0/16 52.113.0.0/16 52.114.0.0/16 52.115.0.0/16 52.120.0.0/16 52.121.0.0/16 52.122.0.0/16 52.123.0.0/16 which looks better but scares me out. Having no auth is it secure to bind so many ranges to MS? Do you use anything else than certificate verification for these calls? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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Dispatcher failover when SRV failover is on?
by Jon Bonilla (Manwe) 23 Feb '23

23 Feb '23

Hi Again, fighting with MS Teams but this time it's a kamailio question. I have MSteams hosts in the dispatcher for OPTIONS and for destination. Acording to MS you need to try sip.pstnhub.foo then sip2.pstnhub.foo and sip3.pstnhub.foo 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 1 2 | sip:sip2.pstnhub.microsoft.com;transport=tls | 0 | 2 2 | sip:sip3.pstnhub.microsoft.com;transport=tls | 0 | 3 I was going to implement dispatcher failover as usual ds_select_domain (alg 8) with failure routes but checking the SRV of the first domain I see it already includes 4 hosts: host -t srv _sips._tcp.sip.pstnhub.microsoft.com _sips._tcp.sip.pstnhub.microsoft.com has SRV record 10 0 5061 sip.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 20 0 5061 sip2.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 30 0 5061 sip3.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 40 0 5061 sip4.pstnhub.microsoft.com. So... If I have dns srv failover enabled in kamailio... what would be the point of the dispatcher failover? In case I only want failover in case 408 or 5XX wouldn't the tm module do it for me? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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What does l_username in UAC do?
by Дилян Палаузов 23 Feb '23

23 Feb '23

Hello, the uac module-registrations have the fields l_uuid - user for Contact: in REGISTER (before @). Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local unique id used to build and match contact addresses.”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local unique user id, e.g.,: 12345678” l_username - Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local username”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local user name, e.g.,: test ”. For me there is no difference between “local unique id” (as for l_uuid) and “local username“ (here). l_domain - used for Contact: in REGISTER (after @), described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “local domain” When is l_username used, what is it purpose? Does it solely serve as possible filter criterion for the RPC commands: uac.reg_info l_username abc uac.reg_enable l_username abc uac.reg_disable l_username abc, and uac.reg_unregister l_username abc Do I see correctly, that via UAC-RPC an unREGISTER can be triggered, but there is no way to trigger a REGISTER over UAC-RPC? Thanks for the clarifications Дилян

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change default 477 response code per case
by Patrick Karton 22 Feb '23

22 Feb '23

Hello , i would like to know if there is a way to change the default 477 response code generated by kamailio in some cases. for example if kamalio generates 477 because its not able to reach a peer i would like to change it to another code before to forward it. Thanks.

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Call History
by Anthony Wittig 21 Feb '23

21 Feb '23

Hello, I'd like to start tracking a call history/log as soon as I receive a call. I'd like the customer to be able to see that there's an incoming call to one of their numbers in real time. I'm thinking of using the RABBITMQ Module to manually emit an event from my main route when I receive an INVITE. Is that a bad idea? What if the broker isn't located on the same host (latency/availability)? I don't want to negatively affect the call processing. I was looking at the Dialog Module and it has three event routes: * event_route[dialog:start] * event_route[dialog:end] * event_route[dialog:failed] It's really close to what I want, but I'd like to capture the dialog before it starts (Unconfirmed or Early). Maybe another option is to scan the Dialog table with an external process and emit events from there. Has anyone had success in creating a realtime call history? Is there a better way to go about doing so? Thanks! -- Anthony Wittig Mango Voice Developer

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Info: Registration to Kamailio World 2023 is open!
by Daniel-Constantin Mierla 21 Feb '23

21 Feb '23

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (June 5-7, 2022, in Berlin, Germany) is now open! The event returns to an in-person conference format, at the same wonderful location in the city center of Berlin! More details are available at: * https://www.kamailioworld.com/k2023/registration/ Call for presentations is also open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

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topos breaking route set on hairpin calls?
by Benoît Panizzon 20 Feb '23

20 Feb '23

Hi Having this situation while elaborating topos in lab. * Kamailio Registration Instance with topos enabled where customers are registered. * Kamailio Core Instance with Dialogue enabled and handling interconnections. Calls in transit from customer to IC work flawlessly, topos does what it should, hides and restores headers needed on the core. But I have run into an issue with customer calling another customer on same registrar instance. To generate a CDR with diaglogue and for decision where to route the calls to (registrar is kept simple and only handles registrations and locations), those calls are also routed via core. This works as long as PRACK is disabled. When PRACK is enabled and CPE B is requesting 100rel, the PRACK send from CPE A has no route set (it was removed by topos). Therefore the PRACK only makes it to the core who then does not know where to route that PRACK. It looks like topos on the REG only restores ONE Route pointing to the core (maybe because the 2nd one pointed to itself?) header when restoring them. I have attempted to spam more record_route() here and there in the config of the core and the registrar to try to force creation of route headers, but failed. Is this a known limitation of using topos? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Re: OT: MS Teams source IP address ranges
by Henning Westerholt 20 Feb '23

20 Feb '23

Hi Jon, you could of course try to add some heuristic, in the end the used IPs are a smaller sub-set. But to just quote from the official docs, it seems to be pretty clear: "You need to open ports for all these IP address ranges in your firewall to allow incoming and outgoing traffic to and from the addresses for signaling." Cheers, Henning ________________________________ From: Jon Bonilla (Manwe) Sent: Monday, February 20, 2023 15:49 To: Henning Westerholt; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] OT: MS Teams source IP address ranges El Mon, 20 Feb 2023 13:06:30 +0000 Henning Westerholt <hw(a)gilawa.com> escribió: > Hello, > > correct me if I am wrong, but the second group of addresses ("/16") is > included in the first group ("/14"), right? > > Regarding the large IP scope, this is the way Microsoft designed it, > unfortunately. There is not that much what you can do about. > > Right. Didn't check the masks. they all were 16 in my mind :) So, if I add those ranges in the permissions table my only way of avoiding spoofed calls is certificate validation? I'm already doing that and also checking the company domain and ms account email address. Just wanted to be sure that nothing "essential" was missing. -- PekePBX, the multitenant PBX solution https://pekepbx.com

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sendig 4 byte udp nat pings
by Bernd Krueger-Knauber 18 Feb '23

18 Feb '23

Hi, maybe I'm stupid, but I can not find out how to send 4 byte UDP NAT packets to the sip server. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 15) modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") I'm a absolute beginner with kamailio and the page about the nathelper Module is not really helpful. At the moment no pings are sendet. But I receive 4 byte packets from the sip server. After 15 minutes my calls are cancelled because of the missing pings. Without proxy, the phone send the pings directly to the sip server and the call stays alive. When I experiment with some other nathelper settings, kamailo start results in errors. Any help would be appreciated. Best regards, Bernd

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