Hey Folks,
What a great conversation, really enjoyed all the insight and feedback, other than what's already been said, which I agree with whole heartedly, I want to add some personal reflection.
FOSS is exceptional for putting tools in your hands to build what your business can use to succeed. You will spend upfront $$ investment with man-hours learning, deploying, managing then cost trails off to realize the return on investment which is usually exponential compared to commercial drop-in voice switches. A key consideration is expected/targeted growth of the solution and operational expediency, how big with the subscriber base get, voice service is a high touch point for customer support. Once you (as in personally) deploy a FOSS solution in production, you are tied to that solution for its foreseeable existence, good luck finding competent and like-minded staff that see your vision and can support the solution as you do. Deployment guys will always have a deeper understanding/knowledge of the solution, architecture, configuration, pitfalls and work around than the operations support folks. If you don't have staff resources, having consultants you can rely on is crucial for success.
I can't say enough about Kamailio for building call routing solutions, mature, stable, reliable, continued improvements, phenomenal developer and community base for support.
Good luck whichever direction you go in!
JR Richardson
On Apr 14, 2024, at 12:25 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote:
> Medium to large organisations, in particular, tend to extract labour from consistent (if unextraordinary) output of everyday do-gooders, and not the more stochastic and volatile heroics of open-source superstars.
Sorry, I meant to say "extract value" here, not "extract labour".
But the larger point is that any successful formula heavily reliant on open-source is going to be a lot more dependent on culture, and on the strengths of individual people and their specific skills, than a formula reliant on a third-party vendor, all other things being equal.
-- Alex
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
End of sr-users Digest, Vol 227, Issue 36
*****************************************
Hi
Using Kamailio 5.6 and trying to get Teams integration working. Failing at the first step! I receive no reply from MS to my OPTIONS ping. AFAIK it is correect, here is the OPTIONs sent:
OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0
Via: SIP/2.0/TLS msteams.domain.com:5061;branch=z9hG4bK0c2b.da69f3c3000000000000000000000000.0
To: <sip:sip.pstnhub.microsoft.com;transport=tls>
From: <sip:msteams.domain.com>;tag=6d9effc2a8ec7e0b16377e93bd9e71e2-470e3c71
CSeq: 10 OPTIONS
Call-ID: 6ea8091201db565b-605531(a)159.65.196.10
Max-Forwards: 70
Content-Length: 0
User-Agent: ABC SBC
Contact: <sip:msteams.domain.com:5061;transport=tls>
From the logs (below) I can see that w are accepting the MS cert OK but get no response and ends with a 408. The domain is configured in Teams, and a user with a license was added. MY Cert is issued by GeoTrust TLS RSA CA G1.
Any idea if the issue might be on my side?
024-04-15T14:53:19.302610+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_domain.c:798]: sr_ssl_ctx_info_callback(): SSL handshake done
2024-04-15T14:53:19.302692+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful
2024-04-15T14:53:19.302762+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:545]: tls_connect(): tls_connect: new connection to 52.114.14.70:5061 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256
2024-04-15T14:53:19.302824+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:550]: tls_connect(): tls_connect: sending socket: 159.65.196.10:0
2024-04-15T14:53:19.302868+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com
2024-04-15T14:53:19.302917+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 03
2024-04-15T14:53:19.470476+00:00 msteams /usr/sbin/kamailio[605606]: DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fd3c77639c8
2024-04-15T14:53:28.770710+00:00 msteams /usr/sbin/kamailio[605591]: DEBUG: dispatcher [dispatch.c:3617]: ds_options_callback(): OPTIONS-Request was finished with code 408 (to sip:sip.pstnhub.microsoft.com;transport=tls, group 1)
-Barry
Hi,
I am using kamailio with KEMI Lua. The script has some medium level
complexity as it stores some information from the invite in redis
cache during the call and uses that information during the call transfers
(REFER and Invite triggered by REFER).
Lua script itself creates/sets some variables, and
sends/retrieves information during the flow for example
KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " ..
KSR.pv.getw("$ci") .. " EX 3600", "r");
KSR.ndb_redis.redis_cmd("srvN", "GET refvar-" .. cid, "r");
reft=KSR.pv.getw("$redis(r=>value)");
I made a test call and disconnected, what I am observing is that even if
the call is finished the used memory is slightly increased.
before:
< real_used: 844240
< real_used: 844088
< real_used: 844088
< real_used: 844088
< real_used: 844088
< real_used: 844088
< real_used: 844088
< real_used: 844088
after:
> real_used: 846080
> real_used: 845864
> real_used: 846584
> real_used: 846304
> real_used: 846200
> real_used: 846120
> real_used: 845248
> real_used: 846168
My assumption is after finishing the call, memory used by variables and
redis ops is released and should be back to original values but its not
happening. In production this issue is causing some problems as I have to
restart the kamailio process after every 3-4 weeks so that it starts
breathing with more available memory.
Initially I thought its Kamailio version issue as in my last posts in this
forum people advised me to upgrade to the latest kamailio version (original
one was 5.4.0) but the same issue is happening on 5.7.0.
Can someone explain how to debug/troubleshoot this situation . I have a
parallel lab environment, I do not see anything interesting in syslogs.
--
Muhammad Danish Moosa
" The core of mans' spirit comes from new experiences. "___ Christopher
McCandless
Hello,
there is some planned work on the data center infrastructure hosting
kamailio.org server that is going to affect briefly the web server (web
site, wiki) and the mailing lists.
The downtime is expected to be 10-15 minutes around 1:00am Western
Europe time (CEST / GMT+2), but if it takes longer or repeats, be
patient for a while.
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com
Hi Everyone,
I would like to hear some stories about moving away from proprietary telephony services whether on premises or on cloud to FOSS solutions and how much stable, secure, efficient and cost effective it was.
I would like to present to my managers such cases to convince them to move away from proprietary telephony as it is just huge amount of cost and technical debt.
Thanks everyone and really appreciate any insights, also I am sorry if this is not the place to ask for such!
hi,
i have kamailio with TLS
i dont have homer and voipmonitor with TLS decryption doesnt work
how can i check final SIP INVITE (after all rewrites/modifications)
before is send to the peer?
tried
onsend_route {
if ($rm == "INVITE") {
xlog("L_INFO", "Final SIP Message before
Sending: $mbu\n");
}
}
but record-route/via are different than reality (checked with sngrep
with TLS disabled)
kamailio log
Record-Route: <sip:;r2=on;lr;ftag=b0d1d4ce-...
Record-Route: <sip:10.10.10.10;r2=on;lr;ftag=b0d1d4..
Via: SIP/2.0/UDP 30.30.30.30:5080;TH=div;rport;branch=z9hG4bK
sngrep
Record-Route: <sip:9.9.9.9;r2=on;lr;ftag=246bdf1c...
Record-Route: <sip:9.9.9.9;line=sr-Jf7BrWSBdWSzrD
Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKdb73.0735168f3e5f....
Via: SIP/2.0/UDP
9.9.9.9;branch=z9hG4bKsr-2y7Ud3XCMD4N6Vi8M-iCM-XGdWSBMDGzuhJZu-iGMh9ztEu7q.....
10.10.10.10 - private kamailio ip (fictional)
30.30.30.30 - ip of SIP PBX (caller)
9.9.9.9 - public kamailio ip
Marek
Hi All,
My requirement is when incoming call comes from internal user in i want to put kamailio ip and port in Contact header and forward or dispatch request to sip trunk carrier but i am unable to do so using KEMI frame work LUA. Can someone please give some pointer please?
KSR.textops.remove_hf("Contact");
KSR.textopsx.assign_hf_value("Contact","<sip:1234@4.5.6.7>");
KSR.rtpengine.rtpengine_manage0();
if KSR.tm.t_relay()<0 then
KSR.sl.sl_reply_error();
end
Hello,
We're using kamailio, with rtpengine for RTP encryption/decryption. It
works fine but I would like to understand the integration better.
How does rtpengine get the TLS certificates, and what crypto library does
it use (openssl?).
Thank you in advance,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782