Hello,
I am planning to create the got branch 5.1 (to be used for v5.1.x
series) by end of next week (likely on Friday, Nov 10, 2017). After
that, the master will be open for new features and fixes will need to be
backported to 5.1.
Should anyone want a different schedule, reply to the mailing lists and
propose your plans.
Cheers,
Daniel
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Daniel-Constantin Mierla
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Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com
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Hi Guys
Please forgive me if I seem to asking stupid questions but I am a complete dummy at the moment regarding this software.
I would love to be able to set up two kamailio servers with the idea of expansion in the future to handle SIP registrations, however having a two server model how would I get the calls to route to another user that is registered on the other kamailio box?
Kind Regards
Daisy
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Hello kamailio folks
i am successfully storing and dumping messages for off line users using
msilo content type text/plain.
i want to store off-line messages for pictures and videos.
my particular queries are
1) is msilo store picture and videos? (i have observed blob in silo
database table which is used for pictures.
2) in case it does what will be declared in content type hdr parameter?
any help in this regard will highly be appreciated.
regards,
sscc
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Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
Alex and Daniel,
Thank you for the feedback, the tip from Daniel helped me pinpoint the
problem.
And the the one from Alex showed that.
The problem was between keyboard and chair, I should have seen that the
SBC removed the PAI.
Once I changed that, the PAI arrived at Kamailio and manipulation was a
breeze.
Thanks guys.
Hi guys.
I'm trying to realize active/active schema with kamailio.
Topology is as following:
active/active kamailio -> routing server (asterisk)
Kamailio servers work as registrar/location servers and sbc.
Asterisk server performs routing functions for calls.
We use sip / sips / ws / wss for signaling.
Schema built on aws platform and both kamailio have virtual ip address.
So when first kamailio crashes, second one assigns ip address of first
kamailio. And vice-versa.
Both kamailio listens on both shared addresses all the time (with help of
sysctl non lolcal bind function).
Problem occurs when first kamailio goes down and second kamailio gets ip
address of first one. Asterisk tries to send in-dialog requests (for
dialogs that were established on first kamailio), request goes to second
kamailio and asterisk gets a socket error.
And this is proper behaviour I think, because first kamailio established
tcp connection with asterisk from certain port, that is not listened on
second kamailio.
So that, I think that the solution is to use force_send_socket function,
but I'm not confident about it.
If anyone has ideas, please send your advices.
Thanks in advance.
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Hello,
I have a problem with kamailio (v4.4), when i try to call from a sip phone through kamailio, the call is interrupted and hang up automatically but after 4s it continues to retry (I can see in the log file Session Progress) and goes through the asterisk (certified/13.13-cert4) and establishes a normal call (It rings at the client side) while it's still disconnected from the sip phone side
But when i try again it works fine sometimes.
Am I missing something in my configuration file? What can I show in order to help me?
Regards
Hi!
I found a few tutorials from Daniel on this topic , the latest being https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astd… .
Has anyone ever tried this with a more recent Asterisk version (13 or newer) and using chan_pjsip instead of chan_sip?
I'd be willing to update the tutorial once I get it running but I would like to avoid going through obstacles that were already handled ;-)
My desired setup is 2 Asterisk boxes (which share configuration, useres and registrations via DB) behind a Kamailio as Load-balancer (got that working already).
With best regards
Florian Floimair
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Hello,
I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster.
I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors:
ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed
ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading
My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file)
[server:default]
method = TLSv1+
verify_certificate = yes
require_certificate = yes
private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem
certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names
ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs
[client:default]
method = TLSv1+
verify_certificate = yes
require_certificate = yes
private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem
certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem
ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem
When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved.
The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate.
As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this?
Thanks,
Francisco.
Hello Kamailions,
I receive anonymous calls where the PAI has a prefix which I need to
strip before forwarding the call to a proxy.
I am trying to read the the header with $hdr(P-Asserted-Identity) and I
tried the $ai pseudovariable, but I get an empty reply back every time.
xlog("PAI is $ai"); results in ; "Nov 1 16:19:55 proxy04
/usr/local/sbin/kamailio[14814]: ERROR: <script>: PAI is <null>"
Same for xlog("PAI is $hdr(P-Asserted-Identity). This works for the
Privacy header but not the P-Asserted-Identity header.
Hope someone can tell me what I am doing wrong....
Rgds,
Gertjan
