Am 25.08.2016 um 16:19 schrieb E. Schmidbauer:
Hi Matt,
Check out this post and config:
https://blog.voipxswitch.com/2016/08/11/kamailio-and-freeswitch-on-the-same-server-with-nsq-and-jansson-rpc/
It demonstrates how to handle NAT & forward a REGISTER to FreeSWITCH
(on localhost)
Thanks,
Am 07.06.2015 um 05:06 schrieb Antonio Gómez Soto:
Hello,
This is question on PBX behavior, what is the right thing to do, and how
do PBX's generally behave.
If a user on a phone, dials a number, which happens to be configured on
the same phone system (for example another tenant), there are
Hi!
I think he refers to IMS where IPsec is used dynamically.
With the first REGISTER the client and the server exchange data which
can be used (together with the password) to calculate the IPsec security
associations. Then the client and the SIP server communicate with the OS
to setup the
Hi!
I just found out that there is a new rtpproxy release:
http://www.rtpproxy.org/post/v2release/
Has anybody tested it and want to share some experiences? Or have people
turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do
not need new features).
regards
Klaus
On 18.05.2015 15:41, Daniel-Constantin Mierla wrote:
Hello,
On 18/05/15 14:57, Klaus Darilion wrote:
Hi!
Should log_prefix also work for xlog in reply routes? In my case it
doesn't (Kamailio 4.2.2)
it should work for received sip replies.
Not here, neither in the default reply route
[dlg_handlers.c:831]: dlg_new_dialog(): Callid
'035a329653a80be399f7b45e6c734415@192.168.149.126' found, must be a
spiraled request
Jan
Klaus Darilion schreef op 2015-01-23 11:38:
I think the problem with 4.2.2. is the following:
After dialog cleanup code is not executed after script exit. Thus
be a spiraled request.
regards
Klaus
On 22.01.2015 21:02, Jan Hazenberg wrote:
Klaus,
Yes, that solves the problem here as well.
Thanks,
Jan
Klaus Darilion schreef op 2015-01-22 16:16:
Hi Jan!
I replaced dlg_manage with setflag(dialog flag). This way the dialog is
created only
Hi!
I use the dialog module to count and limit concurrent calls per user. It
worked fine with 4.1.7 but fails with 4.2.2.
My config basically looks like:
route{
...
dlg_manage()
...
authentication (stateless replies + exit)
...
t_on_reply()
t_on_branch()
...
t_relay()
exit;
}
Nice, I have the same problem after upgrading from 4.1.7 to 4.2.2.
. in my case it seems the dialog callbacks are not executed for
responses ... I am still debugging .
On 22.01.2015 13:25, Jan Hazenberg wrote:
Hi All,
I'm running into a issue with the dialog module since the
Sounds like a bug.
Anyway, as fix_nated_contact is not standard conform, it is better to
use handle_ruri_alias() and add_contact_alias() (see the Kamailio
default config file). Maybe they handle also IPv6 correct.
regards
Klaus
On 22.01.2015 16:26, Sebastian Damm wrote:
Hi,
I'm trying to
Hi Jan!
I replaced dlg_manage with setflag(dialog flag). This way the dialog is
created only when a transaction is created. It solved my problems.
regards
Klaus
On 22.01.2015 14:47, Klaus Darilion wrote:
Nice, I have the same problem after upgrading from 4.1.7 to 4.2.2.
. in my
Hi!
Currently tsilo allows to add branches in failure route or request route
(e.g. on REGISTER).
would be be possible to allow this function also in reply routes? E.g.
if one branch replies with 3xx, to add a new branch to the CFWD target
but still have the other branches ringing.
regards
Klaus
Hi!
Kamailio 4.1.6 with default settings for usrloc and registrar modul and
db_mode=1 (write through).
Fritzbox sends a reREGISTER and Kamailio tries to insert the contact
into the database, but this fails as this contact for this AoR already
exists.
So I think there are 2 scenarios: Kamailio
, Klaus Darilion wrote:
Hi!
Kamailio 4.1.6 with default settings for usrloc and registrar modul and
db_mode=1 (write through).
Fritzbox sends a reREGISTER and Kamailio tries to insert the contact
into the database, but this fails as this contact for this AoR already
exists.
So I think
Hi!
It seems that uac_replace_from does restore the original URI only if the
From URI is changed. Only changing the Display-name, eg:
uac_replace_from($avp(pad),);
does not add the RR-cookie, thus the original will not be restored (e.g.
on responses).
Is this on purpose or a bug?
Further more, auto mode with restore_dlg=1 does automatically change
the From-uri for in-dialog requests, but not the From-display name.
Is this also intended?
Thanks
Klaus
On 10.10.2014 16:02, Klaus Darilion wrote:
Hi!
It seems that uac_replace_from does restore the original URI only
On 02.10.2014 10:48, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
One more came in my mind:
- failure_reply_mode in tm set to 3 (now is 0)
i have been using 3 there.
IMO this makes sense
___
SIP Express Router (SER) and Kamailio
Something is going wrong here:
On 24.09.2014 18:41, Igor Potjevlesch wrote:
DEBUG: rr [loose.c:90]: is_preloaded(): is_preloaded: No
That's correct. The ACK is not pre-loaded (with a route set).
Checking first local URI (either alias= or listen= statement)
DEBUG: core [socket_info.c:583]:
...@lists.sip-router.org] De la part de Klaus Darilion
Envoyé : mardi 23 septembre 2014 16:04
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] Relaying ACK to Asterisk
dump the whole ACK packet received by Kamailio, and the packet looped by
Kamailio
regards
Klaus
On 20.08.2014 18
Hi!
I have to upgrade a production server from 3.x to 4.x.
4.x added a unique constraint on ruid.
I have plenty of entries in aliases and usrloc table. How do I update
these entries to get the ruid field populated?
Can I do an upgrade reusing the entries stored in the usrloc table or do
I have
dump the whole ACK packet received by Kamailio, and the packet looped by
Kamailio
regards
Klaus
On 20.08.2014 18:38, Igor Potjevlesch wrote:
Hello,
I’m having trouble with this scenario (Kamailio and Asterisk are working
on the same server, Asterisk listens on 4060 instead of 5060):
The TLS module uses OpenSSL libraries.
You can force a dedicated algorithm with the standard OpenSSL methods.
If you want to change an algorithm, you have to change OpenSSL.
regards
Klaus
On 10.09.2014 08:09, aawaise wrote:
Hello,
I have downloaded and unzipped kamailio package available
Kazoo seems to be a product name of 2600Hz. If the module is generic,
then I would suggest to name it amqp, which better describe what it does.
regards
Klaus
On 09.09.2014 14:44, Daniel-Constantin Mierla wrote:
Hello,
I see there are some new functions prefixed with kazoo_, using a
On 03.09.2014 03:09, Muhammad Shahzad wrote:
Thank you so much for your informative response.
Yes the peer may be correct term in this sense as i am trying to
identify devices (SIP UAs or Proxy) that are directly connected to
Kamailio via SIP signalling (i.e. there is no other intermediate
Not sure what you trying to do, but the Via header is for transactions.
It may be different for every transaction. Thus, if you need transaction
matching (requests to responses) then you are fine and should use purely
the branch id.
If you want to match messages from one transaction to messages
it. Maybe you can adapt the old
patch if it not something that complex and you have time to look at it.
Otherwise, any further details about what you had to do in the past
would help to add support for it again.
Daniel
On 02/09/14 15:57, Klaus Darilion wrote:
Indeed, currently Kamailio
On 21.07.2014 14:59, Daniel-Constantin Mierla wrote:
Hello,
you may get similar results using t_cancel_callid(():
- http://kamailio.org/docs/modules/stable/modules/tmx.html#idm8272
For each call you have to store the call-id, cseq and the target user
somehow (e.g., using htable).
Answering myself: This is a feature.
Assigning AVPs multiple times add an additional AVP with the same name.
To overwrite an existing AVP the following syntax is needed:
$(avp(pattern)[*]) = .
regards
Klaus
On 07.07.2014 12:50, Klaus Darilion wrote:
Hi!
Kamailio 4.1.4. When using
Hi!
Kamailio 4.1.4. When using avp_check with fnmatch and AVP the result is
always TRUE if it is true for 1 time. It only happens with $avp(), but
not with $var().
Is this a bug or some special behavior of AVPs?
xlog(L_ERR, === with $$avp(...));
$avp(rpid) = +499539110;
It may work catching 200 OK in reply_route, but catching =300 in
failure route. The failure route is executed only once for all branches
(it chooses the most important response code)
regards
Klaus
Am 04.06.2014 11:49, schrieb Sebastian Damm:
Hi,
I have a scenario where I want to send a
Seems like you should an B2BUA/SBC (take a look at sems)
Am 05.06.2014 20:53, schrieb Maciej Marczyn'ski:
Hi Everyone,
recently I'm playing around with the topoh module but I couldn't find
the functionality I'm looking for.
I'd like to hide my network topology before the end-user but not only
you have to configure APT:
http://www.kamailio.org/wiki/packages/debs
On 19.03.2014 10:26, Swetha Raj wrote:
Hi,
I am trying to install Kamailio on Ubuntu 12.04,but hitting the
below issue.
root@test:/usr/local# apt-get install kamailio kamailio-mysql-modules
kamailio-tls-modules
AFAIK Asterisk supports multiple domains, but it seems that is not used
for registration.
On 09.03.2014 00:58, Alex Villacís Lasso wrote:
The above is the non-encoding version. What is the best way to modify it
to do a domain encoding in the username? Or, if a better solution
exists, what is
On 10.03.2014 06:18, Juha Heinanen wrote:
rr module has check_route_param(re) function that can be used to check
if local route header has a param that matches re. however, there does
not exist a function to find out what is the value of a given rr param.
for example, if route header has
Thanks to all involved
On 03.03.2014 14:16, Victor Seva wrote:
The new build system for Debian and Ubuntu packages is now in place.
This service is kindly sponsored by SipWise [0] thanks to Andreas
Granig [1]. Sipwise is providing the hosting and man power to create
and manage this new system.
Am 26.02.2014 22:48, schrieb Henry Fernandes:
Is there a way to prevent relaying requests to myself in route[RELAY]?
To relyable prevent this (eg. the destination IP address (your server IP
address) may be hidden behind others domain) you have to check the
destination in the branch route.
For sure I would use Kamailio as an SBC, but nevertheless I see these
sortcoming in Kamailio (I don't know hov other SBCs handle this)
- config changes require a restart: most of the time this goes fast, but
sometimes processes may fail to start (ports not freed by the OS, ...).
Further,
Am 27.02.2014 23:43, schrieb Alex Villacís Lasso:
Is this setup recognizable as an already-solved problem (minus the
localhost trick)? How is it done correctly?
I think your setup is too complex. If I didn't missed your requirements
I think you can do it this way:
From routing point of
Puh, too many questions in one email.
First, you should describe what you want to achieve. Eg. is there
routing between the networks done by the server? E.g. can a clinet on
10.1.0.0/24 ping a client on 192.168.0.0/16? If yes, there is no need
for Kamailio/Asterisk to listen on multiple
listen=tcp:10.2.3.4:5080
You only have a single socket: TCP.
If you want to use websocket over TLS you need at least also:
listen=tls:10.2.3.4:5090 (or whatever port you like, eg. 443)
Further, if you talk to Asterisk with UDP, you also need a udp listen
statement.
regards
Klaus
I am a bit confused. I suspect with SBC you mean the Kamailio proxy?
On 05.02.2014 14:54, Diego Alejandro Ozuna Escalada wrote:
When a UA sends an INVITE to the SBC, the responses (back) to UA are
being sent with the correct external source IP (X.X.248.194) but if
there are retransmissions of
On 05.02.2014 11:20, Daniel Tryba wrote:
The problem I have with add_contact_alias/handle_ruri_alias is that any
kamailio in the path will parse these hints whether or not it is actually ment
for that kamailio.
handle_ruri_alias should only be used by the last proxy in the chain,
thus by
Read the default configuration to understand how routing within Kamailio
works.
Regarding REGISTER: just remove all the REGISTER handling (and PUBLISH,
SUBSCRIBE and NOTIFY) in the config, then the REGISTERs will be routed
like all other messages.
Then take a look at the PSTN gateway
Asterisk's transcation layer is quite buggy - so it may also be that the
reINVITE with Cseq 103 is a retransmission of a previous transaction
(which was not stopped correctly).
regards
Klaus
On 04.02.2014 08:52, dotnetdub wrote:
Hi Olle,
Just a quick update..
I've gone through this in
properly but no audio
flows between phones.
Now I am in the process of trying to locate where is the problem by comparing
how both files handle the NAT support.
Thank you
- Original Message -
From: Klaus Darilion
Sent: 01/23/14 08:12 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re
For file transfer you need an MSRP relay. AFAIK rtpproxy can not handle
MSRP.
See the MRSP module
http://kamailio.org/docs/modules/4.1.x/modules/msrp.html
regards
Klaus
On 29.01.2014 09:44, Wingsravi R wrote:
Dear Daniel Kamailio'ns
I am working on File transferring feature between two
();
}
}
}
If my problem could be caused by a kamalio miss-configuration could you
please send me an example of configuration that should work with
websockets, rtpproxy-ng-mediaproxy-ng in order to remove one possible
cause?
Thank you.
Best regards,
Mihai M
On Wed, Jan 29, 2014 at 1:31 PM, Klaus Darilion
I have not used MSRP yet. But I guess you can detect MSRP by inspecting
the SDP. USe the functions from textops module or sdpops module to check
if the SDP is a normal call setup or an MSRP session.
regards
Klaus
On 29.01.2014 13:13, Ravi wrote:
Dear Klaus,
Thank you for the reply,
Ya i
Am 21.01.2014 17:33, schrieb John Smith:
The next test has been to comment out the rtpproxy_manage at NATMANAGE function
and to put it both at route[RELAY] and onreply(route) following your post in
this list from January
On 23.01.2014 10:29, John Smith wrote:
Hello Klaus,
I had already two sockets bound each to two independent physical interfaces. I
have added the force_send_socket at each rtpproxy
Just for clarity:
force_send_socket is for near_end NAT traversal of the SIP signaling,
whereas
On 20.01.2014 17:48, Keith wrote:
Hi,
I have a kamailio server which is responding to SIP updates but sending
from the wrong IP. I have multiple IPs in the same subnet on the same
NIC. Is there anyway to say send update back out of the interface it was
received on?
I think that should be
On 21.01.2014 12:27, Fred Posner wrote:
With a patched version of rtpproxy you can advertise your private ip.
http://www.fredposner.com/voip/1457/kamailio-behind-nat/
Aha, nice. Haven't known of this one.
I always specified the adverstised IP address when calling
manage_rtpproxy(). That
On 21.01.2014 13:24, John Smith wrote:
I might be making wrong assumptions regarding this traffic flow. Is that
correct?
That depends on your policy. It is up to you to define how RTP should be
routed. There are basically 2 choices:
a) RTP from clients is handled by rtpproxy:
phone1
Actually, it should work without any NAT traversal done in Asterisk, if
Asterisk communicates never direct with the phones, but only via
Kamailio and rtpproxy. In this case, Asterisk can use private IP
addresses. All the near-end NAT traversal can be done in Kamailio.
regards
Klaus
On
? To
check if the IP is from the outside and then rewrite via rtpproxy_offer in the
NATMANAGE block?
Thank you
- Original Message -
From: Klaus Darilion
Sent: 01/21/14 05:25 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio behind NAT
On 21.01.2014 13:24, John
On 19.01.2014 14:05, Volkan Oransoy wrote:
I am trying to setup an active-active pair of sip proxies. I have
connected two kamailio boxes to the same PostgreSQL database and my
usrloc db_mode is 3. I can see the registration data in the database but
ul show outputs of two device is not same. I
() and
fix_nated_register(). Do not use fix_nated_contact(). Further, apply NAT
traversal only if the clients are directly connected to your proxy
(not if there are other devices inbetween).
On Tue, Jan 7, 2014 at 5:01 AM, Klaus Darilion
klaus.mailingli...@pernau.at mailto:klaus.mailingli
On 02.01.2014 17:00, Jr Richardson wrote:
Would it be prudent to open UDP media ports from Internet to PBX's on
a case-by-case basis, basically white listing media streams or is
there any attack vulnerability with UDP in the media port range or
should I open up media port range to all PBX's
On 03.01.2014 16:59, Brian Davis wrote:
REGISTER sip:test1.test.com:5060 http://test1.test.com:5060 SIP/2.0
Via: SIP/2.0/UDP
96.xxx.xxx.xxx:33745;rport;branch=z9hG4bKf5s1p`n3TRv5TZx5RXy.RVv+JPz8Nat*UX!8KRx4SRx
Via: SIP/2.0/UDP
If rtpproxy is behind NAT, you usually have to instruct Kamailio to
write the public rtpproxy IP address into the SDP, instead of the local
one (which is sent from rtpproxy to kamailio).
e.g: rtpproxy_manage(co,your.public.ip.address);
regards
Klaus
On 20.12.2013 17:07, Benjamin Trent wrote:
FYI: In master there is a nice way to get all the BTs:
utils/kamctl: new command 'trap'
- useful to get a full bt dump of all kamailio processes
- handy in dead-lock investigatigations
regards
Klaus
On 20.12.2013 19:06, Daniel-Constantin Mierla wrote:
Hello,
the bt is from custom timer
Try with a static assignment with force_send_socket(). If this works,
try a static assignment with $fs. If this works, try the dynamic
assignment with PVs.
regards
Klaus
On 11.12.2013 11:32, Keith wrote:
Thanks for the info guys, unfortunately it's not sending the from ip
address properly
On 03.12.2013 14:23, Mark Zeman wrote:
Hello all,
The subject says most of it, I think.
We set up our Kamailio and RTPProxy according to
http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
with the addition of an alias (siplab.ch), and the DNS to go with it, as
well as TLS and SRTP.
That's quite easy - that's a typical load-balancer setup.
Just store the mapping for example in a DB and then use the sqlops
module to query the DB and get the respective IP address of the user.
But before you add this routing logic I would recommend to add
Kamailio with a static forwarding
I think it is a conceptual question indeed. You abuse the 403 error in
some table (actually missed_calls is for missed calls, not for
rejected calls) to log/account a rejected call.
Make it more explicit. If you want to track rejected calls, make a
dedicated table and insert an record into
A hack would be to loop the CANCEL to Kamailio again and forward it then
stateless. When forwarding the CANCEL stateless, you can add headers.
regards
Klaus
Am 20.11.2013 15:19, schrieb Grant Bagdasarian:
I see, so there is no way to append a header to the CANCEL created by Kamailio?
I tried
On 18.11.2013 15:45, Alex Balashov wrote:
http://kamailio.org/docs/modules/4.1.x/modules/tmx.html#idp15326008
Does it only suspend the transaction, but not the script processing?
Is there somewhere a more complete example how to do some async stuff
meanwhile and then resume the
patch was done on 4.0.3.
Thanks,
On 11/15/2013 3:59 AM, Klaus Darilion wrote:
Hi Ding Ma!
It would be great if you can provide the patch at the tracker.
https://sip-router.org/tracker/
regards
Klaus
On 25.10.2013 01:54, Ding Ma wrote:
Is this the right way to build without optimization?
make
happened
even if we reload tls every 5 mins when there are some active TLS
connections. Can we make these fixes into kamailio code base? What's the
process to submit changes for review?
Thanks,
Ding
On 10/24/2013 03:18 AM, Klaus Darilion wrote:
You should build Kamailio without optimizations
I think it would be nice if the CANCELs are sent before the INVITE. But
this will never ensure the order how they are received at the client
side. E.g. there can be packet loss which drops the CANCEL but not the
INVITE, or with load balancing the INVITE can overtake the CANCEL. And
if the
On 25.10.2013 14:19, Vassilis Radis wrote:
Hello,
When I have 2 clients using a kamailio proxy, and both of the clients
are behind their own NAT, then my only options for relaying media
between them is using some kind of intermediate rtp proxy or STUN etc?
STUN is just a method for a client
. The core dump hasn't happened
even if we reload tls every 5 mins when there are some active TLS
connections. Can we make these fixes into kamailio code base? What's the
process to submit changes for review?
Thanks,
Ding
On 10/24/2013 03:18 AM, Klaus Darilion wrote:
You should build Kamailio without
You should build Kamailio without optimizations. value optimized out
does not bring much information.
regards
Klaus
On 23.10.2013 21:48, Ding Ma wrote:
Hi, all
This is related to the previous tls.reload not safe email chain. Now we
have a detailed gdb output that shows the stack trace of the
OF course you could just SUBSCRIBE to get NOTIFYs. But then you would
need to subscribe to all users (e.g. subscribe a user whenever there is
a new registration).
I think a cool feature would be a 'wildcard' subscription, e.g.:
SUBSCRIBE sip:*@mydomain.com to receive all events of
On 21.10.2013 22:47, julian arsanches wrote:
for now i reply with 488 wich tell me something is bad but i thought
that kamailio will know when a message is send from itself to itself.
Sending a message to itself is a legal use case (it is called
spiraling), thus there is no such automatic
On 22.10.2013 08:59, Juha Heinanen wrote:
Klaus Darilion writes:
I think a cool feature would be a 'wildcard' subscription, e.g.:
SUBSCRIBE sip:*@mydomain.com to receive all events of mydomain.com, or
SUBSCRIBE *@*.
yes, cool feature for nsa.
:-)
They don't need this, as still most
Just as a note: If you need to record_route for in-dialog requests then
the clients are buggy and you should report the issue.
regards
Klaus
On 18.10.2013 19:46, Spencer Thomason wrote:
Thanks Daniel,
That did the trick!
BR,
Spencer
On Oct 18, 2013, at 10:32 AM, Daniel-Constantin Mierla
I remember that long time ago there was an email discussing the problem
in details. MAybe it was on one of the old mailing lists (ser, openser).
IIRC the feature and the detailed discussion way by Jan Janak. Maybe
this helps you to refine your Google search.
regards
Klaus
On 19.10.2013
1. you should tell us which line exactly is line 356.
2. Maybe DBURL or MULTIDOMAIN are not defined.
regards
Klaus
On 19.10.2013 23:27, Cory Sanders wrote:
Sorry if this is a duplicate email. First one bounced.
I am having trouble starting Kamailio. Please see below for the error
output
() and record_route_preset().
Cheers,
Daniel
On 10/18/13 8:23 AM, Klaus Darilion wrote:
/
//
// On 17.10.2013 17:31, julian arsanches wrote:
// Hi all, before hand thanks for all the support received on this channel.
//
// I have an issue with an installed server on a aws instance which
On 17.10.2013 17:31, julian arsanches wrote:
Hi all, before hand thanks for all the support received on this channel.
I have an issue with an installed server on a aws instance which is
giving me routing loops, my setup is simple, i have alias set for both
ips internal and external and the
On 15.10.2013 13:21, Coy Cardwell wrote:
Thanks.
I am using DB only mode.
There will be a number of servers in the end, so i will have to look
further into the issue I guess, since assumptions were made about how
Kamailio works.
From what I can tell, I think I will have to implement the
Hi Coy!
On 16.10.2013 14:29, Coy Cardwell wrote:
Thanks.
By as long as IP connectivity between the outbound proxies and
registrars is not filtered, what exactly must not be filtered?
The proxies and their 'local' registrars will be in the same private IP
cloud.
Then it should be fine.
Are
Hi!
First, if 2 servers share the same DB, then userloc must be switched to
DB-only mode:
http://kamailio.org/docs/modules/4.0.x/modules/usrloc.html#idp16939424
But this leads you to another problem. As Fred already mentioned, SIP
clients (or the NAT of the user) often refuse messages which
On 14.10.2013 14:57, Keith wrote:
Hi,
Klaus, thank you for pointing me in the right direction with SIP trunks,
got it working so thanks! Basically I did exactly what you said:
- Dialled number
- Match that number to a registered user (had to create a new table for
that)
- Lookup user
-
To dynamically add routes you do not need to understand how the code
works. Use one of the routing modules (lcr, drouting, prefix_route,
carrierroute ). Ususally they are configured via a DB backend and
perform a DB lookup for every call, or cache the routing table and the
routing table
Am 09.10.2013 17:56, schrieb Keith:
Hi,
Can anyone point me in the right direction for setting up SIP trunks?
Whenever I send a call to a registered user on a trunk it just sends
to destination s@x.x.x.x. Is there anyway to say these extensions are
location at this destination IP and port.
Am 08.10.2013 12:14, schrieb Grant Bagdasarian:
Hello,
I've setup two Kamailio machines, one which does all the processing
and the second one which always replies with a 500 Server Internal
Error, to test my Dispatcher fail-over.
When routing a call, the call is always routed to the second
Hi!
keep_hf() keeps all headers, as every header matches the regexp .
IMO an empty regexp should remove all headers (except the mandatory
ones). Otherwise I have to use something like
keep_hf(hope-this-header-never-exists)
regards
Klaus
___
SIP
Use append_to_reply(txt) before sl_send_reply():
http://kamailio.org/docs/modules/4.1.x/modules/textops.html#idp17040608
regards
Klaus
On 03.10.2013 11:38, Grant Bagdasarian wrote:
Hello,
Is it possible to append a new header to a reply generated by Kamailio
and also have it present when
Do you need the regex?
Does remove/append_hf work? eg:
remove_hf(X-Dispatcher);
append_hf(X-Dispatcher: $(avp(dsattrs){param.value,dispatcher}));
regards
klaus
On 02.10.2013 15:55, Grant Bagdasarian wrote:
Hello,
I’m trying to replace the value of a custom header in the failure_route,
Hi!
Question to the experts: Is keep-alive for the Websockets TCP connection
automatically done by the Websockets Layer (client or server), or do I
have to do it manually (nathelper pinging).
Thanks
Klaus
___
SIP Express Router (SER) and Kamailio
a look at the keepalive_.* modparams for the websocket module.
The TCP connection timeout should be set to something a little greater
than the WebSocket ping interval is set to.
Regards,
Peter
On 26 September 2013 12:37, Juha Heinanen j...@tutpro.com
mailto:j...@tutpro.com wrote:
Klaus
If you want to hide your upstream providers from your customers, you
should use a B2BUA (sems, Asterisk).
Probably the most simple setup would be:
customers -- Kamailio -- Asterisk -- PSTN Provider
Kamailio is used for Registrations, NAT traversal, Authentication ...
Calls to
Kamailio uses OpenSSL. Thus, it mainly supports what your OpenSSL
version supports. This means, Kamailio can use all TLS versions and
ciphers that your libssl supports.
You can configure Kamailio to use certain ciphers, see:
http://kamailio.org/docs/modules/4.0.x/modules/tls.html#cipher_list
I also saw this recently in Kamailio and can not remember having this
seen before.
In your case Kamailio is doing some loose-route/strict-route conversion
which of course is buggy.
I think the problem is related that Kamailio does not correctly identify
that the Route header addresses
, Ziad Habchi wrote:
Dear Dragos,
Did you get my email ? any updates regarding the below?
Regards,
Ziad Habchi
-Original Message-
From: Ziad Habchi [mailto:ziad.hab...@jinnysoftware.com]
Sent: Thursday, August 8, 2013 10:38 AM
To: 'Dragos Vingarzan'
Cc: 'Klaus Darilion'; 'Kamailio (SER
It seems that there is a NAT-ALG between the client and the proxy which
rewrites the VIa header and Contact header. These ALGs are quite often
buggy and should be avoided. Further, as the ALG rewrites the IP
addresses, it may be that your configuration does not detect the client
as NATed and
On 07.08.2013 13:42, Ziad Habchi wrote:
Hi ,
I managed to run Kamailio to replace OpenIMSCore. I am using boghe
client to sign in, when I do so , my registration is forwarded as per
the trigger point to my AS server.
As I notice , the REGISTER request miss Content-Type header which is
Inspect the received ACK (log $mb) and check the RURI, and Route
headers. Maybe the show some bugs. Also, if you manually apply routing
to in-dialog requests (e.g. forcing the send socket) make sure to not
make mistakes.
regards
Klaus
On 12.07.2013 17:52, hiro wrote:
hi
I have set up tls
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