On Tue, Mar 03, 2020 at 10:44:32PM -0500, Joli Martinez wrote:
> We have several SIP domains coming into our SBC. I need to build a
> Kamailio box that will 302 the call to the appropriate server based on the
> domain. If the call does not come in with a recognized domain I would like
> for it to
On Tue, Feb 11, 2020 at 08:03:29AM +0900, mayamatakeshi wrote:
> Actually, my idea was to do something like this before handing the REGISTER
> to registrar save() function:
> remove_hf("Expires");
> append_hf("Expires: 200\r\n");
> However, this didn'
On Wed, Jan 08, 2020 at 03:52:09PM +, Duarte Rocha wrote:
> However, when "user=phone" is present, Kamailio's parser works differently.
>
> Without "user=phone" -> {uri.user} is equal to
> "+49234598765;npdi;rn=+49-D123"
> With "user=phone" -> {uri.user} is equal to "+49234598765" .
>
> I rea
On Mon, Dec 30, 2019 at 12:04:47PM +, Duarte Rocha wrote:
> How sould a call duration be calculated?
>
> Let's say the call creation on Kamailio as a proxy has those steps :
>
> 1 - Invite is received
> 2 - Provisional responses
> 3 - 200 OK is received
> 4 - ACK to 200 OK is received
> 5 - B
On Tue, Dec 24, 2019 at 12:46:53PM +, Mahesh Kumar wrote:
> Hi Team,
>
> Getting error in auth modules during Kamalio 5.2 installation and doing SIP
> routing, anyone know this why i am getting below error..??
The error is pretty self explanatory:
> config file /etc/kamailio/kamailio.cfg, l
On Thu, Dec 05, 2019 at 10:38:10PM -0600, Yanko wrote:
> I have two virtual machines with two kamailo domains
>
> VM1: Domain: 172.16.16.1
...
> VM2: Domain: 172.16.16.2
...
> In each domain I can make calls. For example, user 2000 can call 2001. User
> 3000 can call 3001.
>
> How can I make call
On Thu, Dec 05, 2019 at 09:37:51AM +, Laurent Schweizer wrote:
> Hello,
>
> I already see old post about this :
> https://opensips.org/pipermail/users/2014-November/030451.html
>
> but I???m interested to know if now they is a solution
>
> so the issue is a RE-INVITE rejected (415 Unsup
On Wed, Dec 04, 2019 at 02:25:33PM +, Hamid Hashmi wrote:
> Can the following call flow be implemented, if yes, can you give some
> suggestions with modules information?
[RBT call flow]
Kamailio itself cannot AFAIK. You need to have something that plays
audio. Maybe it can be implemented w
On Thu, Nov 28, 2019 at 03:21:21PM +, David Villasmil wrote:
> Thanks Alex,
>
> Do you know how to set it? First time doing thousands of tcp registrations..
>
Depends on what init system you use. If you have to ask I pesume you are
using systemd (since they like to reinvent the wheel). The c
On Tue, Nov 26, 2019 at 10:57:57AM -0500, Daniel Greenwald wrote:
> Yes I've played with topoh but I need more than hiding, I need to store
> data (freeswitch's ip) in the contact header.
That can be done with set_contact_alias / handle_ruri_alias from the
nathelper module. But I guess you are awa
On Tue, Nov 26, 2019 at 10:32:59AM +0100, Daniel Tryba wrote:
> Well, the problem happened to me on 2 different loadbalancers (withing
> 24 hours where the loadbalancers had a near identical uptime) For about
> 35m no new connections can be established. Already established
> connectio
On Thu, Nov 21, 2019 at 07:49:28PM +0100, Jose Fco. Irles Dur?? wrote:
> Thanks for the info!
>
> Tomorrow I'll test it with the 5.1.9 version.
>
Well, the problem happened to me on 2 different loadbalancers (withing
24 hours where the loadbalancers had a near identical uptime) For about
35m no
On Sun, Nov 24, 2019 at 07:35:47PM +0600, Sujit Roy wrote:
> How can i change c= and o= in SDP using RTPProxy ?
You must have read over it in the documentation:
https://www.kamailio.org/docs/modules/stable/modules/rtpproxy.html#rtpproxy.f.rtpproxy_ofrer
Thus by using the c and/or o flags your q
On Thu, Nov 21, 2019 at 02:46:39PM +0100, Jose Fco. Irles Dur?? wrote:
> Hi
>
> I have a kamailio 5.1.2 as load balancer and registration offloading,
> but I have a problem with the max tcp connections that it can handle.
> I suspect that is a linux limit, but I don't find the reason or config.
>
On Fri, Nov 08, 2019 at 02:43:57PM +0200, Konstantinos Merentitis wrote:
> Right, apparently the incoming trunk sends a CANCEL request approx 20secs
> after the initial INVITE, i suppose because the kamailio subscriber is not
> answering, so the flow in my logs is normal.
> I should now rephrase
On Thu, Nov 07, 2019 at 04:53:43PM +0200, Konstantinos Merentitis wrote:
> I tried to modify the failure route in order to also forward not answered
> (failed?) calls to asterisk mailbox but i get the following error and the
> call is never forwarded:
You appear to do the same thing I do, except
On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote:
>
> Hm... Maybe there is other module to achieve such functions?
> Best if it would be with cache :)
> But if no - regex also fine.
But if it works it works! I don't think there is a specific module to do
this but it is easy to imple
On Tue, Oct 29, 2019 at 11:24:34AM -0400, Sergiu Pojoga wrote:
> Hi Daniel,
>
> In an ideal world, DNS is, as you said, the easiest of approaches one can
> think of. However, not all endpoints support [properly] SRV failover.
And that is why you do both. Both a hot standby and multiple active
fro
On Mon, Oct 28, 2019 at 06:03:09AM +0100, Youssef Boujraf wrote:
> I am looking for failover service of kamailio servers looks like
> haproxy but udp & tcp.
>
> -??one domain name : sip.secure.com pointing my public ip address.
> - two kamailio server pointing same database content (user, whitelis
On Wed, Oct 16, 2019 at 08:51:35AM -0700, Jasen Hall wrote:
> I have tried an embarrassing variety of weights thinking I was
> misinterpreting the instructions. I've tried weight divisions of: 1/3,
> 10/30, 90/9, 90/10,100/0, 90/30, and more; all with the hopes of generating
> any variance in a str
On Tue, Oct 08, 2019 at 11:07:44AM -0400, PICCORO McKAY Lenz wrote:
> i have the code with an exit, i dont know if are correct that "exit"
> in that line? or not? help me please?
They are correct (to me). After calling www_challenge() you want to stop
any further processing. Same for your 403 cond
On Sun, Sep 29, 2019 at 11:03:47AM +0300, Olli Attila wrote:
> modparam("dialog", "profiles_with_value", "concurrent_calls")
> modparam("dialog", "dlg_flag", 4)
> modparam("dialog", "db_url", DBURL)
> modparam("dialog", "db_mode", 1)
Have you tested with a lower value of
https://kamailio.org/docs
On Fri, Sep 20, 2019 at 03:00:34PM +, Henning Westerholt wrote:
> > expires: I have it set to just before the end of the current unix time,
> > 2018-01-01 00:00:00
>
> Hello,
>
> you mean probably end of unix time at 03:14:07 UTC on 19 January 2038.
Doh, yes :)
__
On Fri, Sep 20, 2019 at 04:37:20PM +0300, George Diamantopoulos wrote:
> Am I right in thinking that merely INSERTing respective rows in kamailio's
> location DB backend will do the trick? Is there anything I need to worry
> about, like contacts being periodically purged? If yes, how do I prevent
>
On Mon, Sep 16, 2019 at 02:06:43PM +0200, Arnout Van Den Kieboom wrote:
>
> If you need more info, feel free to ask!
>
Start looking at the SDP contents involved. What are the rtp endpoints
in the them in the scenarios you are having problems with (the ipadress
in the c= line and the ports in th
On Wed, Aug 28, 2019 at 05:01:28PM -0400, Aingaran Thirunadarajah wrote:
> Hello,
> Newbie here!
> Have Kamailio set up on a 512MB VPS. Created a couple of users, registered
> with Zoiper and can call between them.
>
> I am looking for documentation to connect the server to other VoIP
> providers
On Tue, Aug 20, 2019 at 10:22:26AM -0400, Travis Ryan wrote:
> What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need to
> still register the trunk from each Asterisk box "thru" the Kamailio proxy,
> etc?
>
> Also, I'm merely accepting outside calls and then validating the caller
On Wed, Aug 14, 2019 at 02:52:45PM -0400, PICCORO McKAY Lenz wrote:
> > In my setups I have a limit of 64 requests per 2s. But I also have
> > whitelist (with/via the permissions module) for known high traffic
> > ipaddresses. Dimensioning the pike module for the known high traffic
> > hosts kind o
On Wed, Aug 14, 2019 at 08:47:02AM -0400, PICCORO McKAY Lenz wrote:
> you said: " A simple SIP phone will only send a couple of messages per
> second"
>
> so if i have that special case with dinamyc ip in clients.. who could be
> better to not confuse those clients with intents of attacks?
I'm no
On Tue, Aug 13, 2019 at 03:57:36PM -0430, PICCORO McKAY Lenz wrote:
> # this it's my setup for pike due the dinamyc ip and devices over the
> internet:
> modparam("pike", "sampling_time_unit", 4)
> modparam("pike", "reqs_density_per_unit", 80)
> modparam("pike", "remove_latency", 60)
> ...
> route
On Thu, Jul 04, 2019 at 10:48:02AM +0200, Federico Cabiddu wrote:
> Actually you can force the TCP socket (e.g. sending from the same socket
> you are listening on) if the kernel has support for SO_REUSEPORT (linux >
> 3.9, FreeBSD, OSX) and you enable tcp_reuse_port in kamailio configuration (
> h
On Wed, Jul 03, 2019 at 06:38:28PM +0200, Karsten Horsmann wrote:
> any one here that can imagine why force sendsocket generates an udp packet
> if the target accept only tcp? And without fs it generates an tcp packet.
> For uac registrations outbound?
Reading the cookbook documentation of force_s
On Mon, Jun 24, 2019 at 10:52:36AM +0300, Amar Tinawi wrote:
> Thank you Karsten
>
> i'll give a try
Karsten gives an excelent howto, but personally I use aptitude for these
situations. It gives you a list of all available package versions and
helps you with the conflicts that might surface.
On Mon, Jun 17, 2019 at 05:18:42PM +0200, Laura wrote:
> ++--+--+-+-++
> | Field | Type | Null | Key | Default |
> Extra
> ++--+--+-+
On Wed, Jun 05, 2019 at 01:18:32PM +0300, wrote:
> When User and Group = kamailio I can't start kamailio.service at all. I get
> errors ()??
>
> >?? 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR:
> >[core/tcp_main.c:2855]: tcp_init(): bind(9, 0x7fb6ef0
On Tue, May 21, 2019 at 03:57:02PM +0200, Benoit Panizzon wrote:
...
> So if the Voice Switch is sending back the Register Contact in the
> INVITE, the PBX cannot use this field to determine which extension to
> ring.
>
> So it has to use the To: Header.
>
> Well, not in a forwarding scenario, th
On Thu, May 16, 2019 at 12:37:50AM +1000, Rhys Hanrahan wrote:
> Hi Daniel,
>
> Thanks for this, much appreciated. Was worried this approach was too much
> of a hack and not the right approach - but knowing someone else has gone
> this way gives me confidence it's a reasonable solution. Will try t
On Wed, May 15, 2019 at 09:38:29PM +1000, Rhys Hanrahan wrote:
> Just to add I've also tried adding multiple alias= definitions, but have
> the same issue - kamailio says "user@fqdn" not found in usrloc when doing
> lookup(). Maybe I need to modify my lookup() call to use a hardcoded URI?
> But I h
On Thu, May 09, 2019 at 10:15:16AM +0100, Mark Boyce wrote:
> We???ve been asked a few times recently if we can do screen-pop of
> incoming calls (SalesForce CRM / Zoho Support) so that customers
> details pop up on the display as calls are delivered. Similarly
> ???click to dial??? from such syst
On Fri, May 03, 2019 at 02:39:10PM +, Ali Taher wrote:
> I'm new to kamailio. I plan to setup a sip server cluster, Does
> someone can give me some suggestions if I can use kamailio as front server,
> which handle sip message and bypass rtp media
> message to backendserver, this mean the kamail
On Fri, Apr 19, 2019 at 11:38:50AM +0300, Yu Boot wrote:
> Added this before final "return", it still allow to call from any IP without
> registration. :(
>
> > if(!allow_source_address() || $au==$null)
> > {
> > sl_send_reply("403","Go away!");
> > }
This code is in no way related to registr
On Fri, Apr 19, 2019 at 08:52:30AM +, David Dean wrote:
> I'm using the following rtpengine_offer() to force the use of ICE relay and
> also replace o= and m=
> ?? ??rtpengine_offer("replace-origin replace-session-connection
> ICE=force-relay RTP");
>
> The SDP is being updated to include an
On Fri, Apr 19, 2019 at 09:44:14AM +0300, Yu Boot wrote:
> Following code snippet from default kamailio.cfg never gives 403 if you
> smart enough to set "fromdomain" parameter on Asterisk to Kamailio's IP. How
> to fix it? I want password-based registration (which is OK now) and permit
> calls via
On Mon, Apr 08, 2019 at 05:40:30PM +0530, vinod mn wrote:
> I have a cloud server, when I make call from a sip phone (registered with
> kamailio),
> in the INVITE header I am seeing the via header with public IP, is there
> any way that I can modify via header to send only the private IP.
> The cal
> To answer my own question:
>
> with
> modparam("acc", "time_mode", 2)
> and
> modparam("db_redis", "keys", "acc=entry:callid,time_attr&cid:callid")
> you'd get a millisec resolution timer in the key for redis
Well scratch that, I wasn't looking correctly at the logs. This doesn't
work.
_
> What does work is "time" instead of "time_hires". Since I have no idea
> what values are available (I guess the table names?) and time only has a
> 1 sec resolution, what is a correct version of time_hires?
To answer my own question:
with
modparam("acc", "time_mode", 2)
and
modparam("db_redis
Under
https://www.kamailio.org/docs/modules/5.2.x/modules/db_redis.html#db_redis.sec.usage
there is an example for using db_redis for accounting. This doesn't
appear to work (5.2.2)
modparam("db_redis", "keys", "acc=entry:callid,time_hires&cid:callid")
results in:
db_redis [redis_dbase.c:1886]: db
I have 3 rtp backends defined with equal weights (33 each). But when I
look at the number of calls being handled the spreaded load is always
5:4:3 for the machines in the orderd listed:
modparam("rtpengine", "rtpengine_sock", "udp:10.235.32.60:7723=33
udp:10.235.32.59:7723=33 udp:10.235.32.58:7723
On Fri, Mar 15, 2019 at 02:05:24PM +0100, Olivier wrote:
> I'm looking for a solution to integrate legacy devices to a SIP network.
> More precisely, I need to forward to and receive Clearmode RTP traffic (see
> [1]).
>
> 1. Do you know any Kamailio-compliant RTP engine (rtpproxy, rtp engine, ..)
On Fri, Mar 15, 2019 at 12:37:16PM +0200, Vitalii Aleksandrov wrote:
> Oh, it actually does. If you use ICE=force, rtpengine removes all ICE
> candidates and inserts its own and both call participant can't to talk to
> each other directly but still can use ICE to establish media streams to
> rtpeng
On Thu, Mar 14, 2019 at 06:01:41PM +0200, Vitalii Aleksandrov wrote:
> > What is wrong with the default behavior? That adds ICE records and
> > rewrites SDP c=.
> When a call goes through multiple proxies and every proxy inserts itself SDP
> becomes really huge. What I like in "force-relay" is that
On Thu, Mar 14, 2019 at 02:47:24PM +0200, Vitalii Aleksandrov wrote:
> Well it's mostly rtpengine question but didn't know where should I send it
> and probably the answer will be more or less useful for kamailio users.
I get the feeling. You could open an issue in github, but
sipwise/rtpengine de
On Wed, Mar 13, 2019 at 06:01:39AM +0800, Isravel Raja Thangamani wrote:
> What I want is, I want to register the Kamailio Trunk in an asterisk with
> Username Password authentication,
>
> My current setup for making that,
>
> VoIP Provider -> Kamailio(Public) -> Asterisk(Randomly Changing Public
On Tue, Mar 05, 2019 at 04:22:12PM +, Sergio Charrua wrote:
> Content-Length: 262
>
> v=0
> o=root 1219665045 1219665045 IN IP4 55.66.77.88
> s=SomeSIPGateway
> c=IN IP4 55.66.77.88
> t=0 0
> m=audio 14326 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-e
On Fri, Mar 01, 2019 at 08:23:11AM -0600, JR Richardson wrote:
> My mind is not right on this one, need a pointer. Here is the simple scenario:
> Carrier> The end device is a static IP, the proxy dips database and knows where
> to send numbers destined for the end device, but if there is no
> respo
On Thu, Feb 28, 2019 at 09:03:30AM -0400, PICCORO McKAY Lenz wrote:
> hih thanks for your respond, but seems you dont paid attention to my
> problem, with ports opened and redirected to pulbic ip with the AWS
> firewalling (tech support) call have sound, but and later NAT traversal
> with rtpproxy
On Wed, Feb 27, 2019 at 04:04:45PM -0400, PICCORO McKAY Lenz wrote:
>
> N]OTE: the public ip are not a real interface in the kamailio/rtppropxy
> machine, are provided by the service AWS at amazon! a NAT kind i guess!
>
But how are you calling rtp(proxy|engine) from kamailio? I think you
need to
On Thu, Feb 21, 2019 at 02:59:03PM +0100, Jan-Hendrik D??rner wrote:
> I would like to get parallel forking working on my Kamailio installation, but
> I have trouble to accomplish that.
[serial forking!]
> Is there any working Kamailio example-config, where I can actual see the
> parallel forking
On Thu, Feb 21, 2019 at 04:55:40PM +0530, Prabhat Kumar wrote:
> When i try to login via SIP Client(zoiper) i am getting the following error.
>
> Feb 21 11:20:26 ip-10-0-0-121 /usr/local/sbin/kamailio[21131]: INFO:
> [core/parser/parse_fline.c:144]: parse_first_line():
> ERROR:parse_first_line: m
On Fri, Feb 15, 2019 at 07:23:07PM +0100, Cristian Livadaru wrote:
> You are AWESOME!
> That's exactly what happened, the call went to first asterisk, that one sent
> it further to the client trunk which responded with SIP 500, kamailio sent
> it to the next one, sip 500 again and thus blocking bot
On Fri, Feb 15, 2019 at 05:13:21PM +0100, Cristian Livadaru wrote:
> Hi,
> I have a Kamailio running as Load Balancer and it works great but since a
> couple of weeks I kept noticing 404s in Homer and when looked into it they
> came from Kamailio.
...
> The two asterisks behind it seem fine and I c
On Wed, Feb 13, 2019 at 04:03:52PM +, Jesse Strahn wrote:
> I have a server running Kamailio with dispatcher. I am trying to
> direct calls from kamailio to an SRV record but am receiving a 404
> error on the calls. If I instead place one of the hostnames from the
> SRV record in my dispatcher.
On Mon, Feb 11, 2019 at 09:07:04AM -0800, Julien Chavanton wrote:
> Hi Marco, not sure if it is the same issue, but I am looking at a problem I
> am facing where in-dialog requests are failing after 3 minutes.
>
> It seems you are also using topos_redis
>
> Tracing TOPOS traffic is seems some leg
On Fri, Feb 08, 2019 at 11:08:03AM -0800, Julien Chavanton wrote:
> The solution that worked for me was to use :
>
> trace_mode=1
>
> This is capturing both version of the message, I think this is about using
> a core event hook instead of a transaction callback
Although this is not the solutio
On Mon, Feb 04, 2019 at 01:33:32PM +0100, Victor Seva wrote:
> Hi there,
>
> Is in my TODO list to integrate aptly into our deb build environment.
> https://github.com/sipwise/kamailio-deb-jenkins/issues/9
Okay, good to know that others see this as a "problem". I'll wait
patiently for this enhanc
On Fri, Feb 01, 2019 at 05:44:45PM +0100, Enrico Bandiera wrote:
> Hello, going back to 5.2.0 is actually not possible anymore if you didn't
> save locally the distro packages because right now on the repos only
> Kamailio 5.2.1 is available
I made the same conclusion with an upgrade to 5.1.6 from
On Thu, Jan 31, 2019 at 12:01:26PM +, YASIN CANER wrote:
>
> Kamailio gave this error that couldnt bind tcp connection other side.
>
No idea, what might be wrong with you conf. But you might start by
making a packetdump (of all traffic except eg ssh on all interfaces) to
see if the problem i
On Fri, Jan 25, 2019 at 10:19:45AM -0300, Marcos Pytel wrote:
> Could you help me to catch the coredump file?
Please keep the mailinglist in the loop.
> When I execute bt full the systems shows "no symbols". What package i need to
> perform it?
I don't know what the sipwise config is, but it if
On Thu, Jan 24, 2019 at 02:30:06PM -0300, Marcos Pytel wrote:
> I'm using Kamailio version: kamailio 5.1.6 (x86_64/linux).
>
> When I enabled topos, after a few minutes i get this erros in the log file
> and the LB Service goes down.:
>
> Jan 24 13:20:21 sipwise lb[30130]: ERROR: ndb_redis [redis
On Tue, Jan 22, 2019 at 12:08:34PM +0530, Prabhat Kumar wrote:
> Is there any function for username authentication? as we have for password
> i.e. pv_www_authenticate(realm, passwd, flags)
I guess your only option for now is to use KEMI and your favorite
scripting language available there.
> Als
On Mon, Jan 21, 2019 at 06:55:32PM +0530, Prabhat Kumar wrote:
> How can i modify username before consuming credentials set in *$au*
> variable. I tried $au=$_s("xyz:"+$au); but it says "*read only pvar in
> assignment left side*"
> ERROR: bad config file (1 errors)
$au is readonly. My workaround
On Tue, Jan 15, 2019 at 03:57:54PM +0100, Daniel-Constantin Mierla wrote:
> some off-topic remarks, maybe you can do something about or others can
> confirm/infirm what I am seeing:
>
> ?? 1) I am subscribed to sr-users mailing list with two email accounts, I
> get your email on my secondary accou
On Tue, Jan 15, 2019 at 03:20:10PM +0300, Soltanici Ilie wrote:
> OK, that looks interesting - and I think I would able to generate such
> options from kamailio??- but how do??I?? measure the time for a response for
> this request?
> Is there any variables which can provide response time for some
On Sun, Jan 13, 2019 at 10:08:31PM +0300, Soltanici Ilie wrote:
> With Asterisk, we are able to get some peer round-trip connection statistic
> by setting qualify=yes for the specified peer.
> It sends periodic OPTIONS to the peer and calculates the time round trip time.
> It's something like - "
On Thu, Jan 10, 2019 at 08:53:43AM +0100, Jos?? Antonio Guti??rrez Delgado
wrote:
> Hello, what I need to know is when that user has been registered for the
> last time or, if possible, all the times he has been registered.
>
> If you could tell me how the RPC commands are used, I would be very
>
On Wed, Jan 09, 2019 at 05:30:52PM +0100, Jos?? Antonio Guti??rrez Delgado
wrote:
> Hi, I would like to know if there is a possibility to see a user's
> connection history, or at least their last connection. Thank you
Define connection?
If you want to see where INVITEs are coming from see the di
On Tue, Jan 08, 2019 at 12:08:23PM +0100, Daniel-Constantin Mierla wrote:
> > Yes. I initially had it enabled but later disabled it for testing with a
> > drop in the event_route and never removed it. In 5.1.3 there was no
> > problem, the moment I updated to 5.1.6 the segfaults began and continued
On Tue, Jan 08, 2019 at 11:40:56AM +0100, Daniel-Constantin Mierla wrote:
> > My crashes don't appear to be related, see
> > https://github.com/kamailio/kamailio/issues/1784
> > Mine are triggered by topos/redis, even though I have topos disabled
> > with an even route.
>
> I requested some extra
On Mon, Jan 07, 2019 at 04:21:35PM +, Floimair Florian wrote:
> This turned out to be unrelated to Kamailio itself in our case.
> The problem was that the systemd-journald of the systemd version shipped with
> Debian stretch was sometimes eating up our CPU time on a single-core VM.
> After upg
On Thu, Jan 03, 2019 at 07:06:51PM +, Duarte Rocha wrote:
> As far as i can tell, i have 3 ways to permanently change the memory
> settings in Kamailio :
>
> /etc/init.d/kamailio , /etc/default/kamailio and src/core/config.h.
>
> What's the priority between them? If i have different values in
On Wed, Jan 02, 2019 at 03:25:55PM +0100, Daniel-Constantin Mierla wrote:
> However, there was no follow up, Florian said he has to monitor after doing
> some fixes on the system and see how it goes. Since then I haven't see
> another update, so if you can get gdb backtrace, we can see if it is
> r
On Tue, Oct 09, 2018 at 01:09:46PM +0200, Daniel-Constantin Mierla wrote:
> Hello,
>
> does it happen that you have the pcap with the sip trace for this call?
> If yes, can you send it to me (can be sent directly if you have some
> sensitive data there)?
>
Is there any progress on debugging this
On Wed, Dec 26, 2018 at 06:08:20PM +, Wilkins, Steve wrote:
> Thank you, I just was not sure what else would cause the relayed packets to
> not be sent out to my fios router. As mentioned, I can pick any other server
> in my network and I can see, in the pcap file, that the relay is attempte
On Wed, Dec 05, 2018 at 09:40:38AM +0100, Kjeld Flarup wrote:
> Yes, the Phones may be on either local LAN (Wifi) and Internet via mobile
> data.
How about use different local address, 1 with an advertise for external
clients, 1 without. Have local DNS resolv to the 1 ip without advertise.
__
On Thu, Nov 29, 2018 at 02:32:02PM +0300, Soltanici Ilie wrote:
> We have a Kamailio Instance running on Public IP Address, one of our ISP
> cannot send ACK back to us because we are sending 100 Trying - without
> "received" parameter.
> Is there any way in Kamailio to force a "received" paramete
On Wed, Nov 21, 2018 at 06:24:22PM +0100, ybouj...@by-research.be wrote:
...
> xlog("NATMANAGE coei\n");
...
> Is it an issue with the public ip address configured in the rtpproxy
> (/etc/default/rtpproxy)?
There is no issue. Please take a look at
https://www.kamailio.org/d
On Tue, Nov 20, 2018 at 10:41:51PM +0100, ybouj...@by-research.be wrote:
>
> WHEN I MAKE AN INVITE FROM 801 TO 803, I HAVE THIS ERROR MESSAGE :
>
> KAMAILIO SERVICE (ERROR:
On Mon, Nov 12, 2018 at 10:11:06PM +0100, Henning Westerholt wrote:
> > What is the difference between topoh and topos?
> > I am using topoh which works fine without DB and only needs the mask_key to
> > be the same on the nodes.
> > Is there a benefit for using a DB and the topos module?
> > [..]
On Mon, Nov 12, 2018 at 09:59:10AM +0100, Jos?? Antonio Guti??rrez Delgado
wrote:
> Hi, I would like to know if it is possible to make the last register of a
> user the active register and if possible the only one.
> I'm with an Android application and sometimes I can not deregister properly
> bef
On Thu, Nov 01, 2018 at 09:20:46AM +, Toffi Bossol wrote:
> my use case will be:
>
>- We use two Kamailio instances (A and B)
>
>- A Client registers to a Kamailio A using TLS (SIP over TLS). The TLS
> session data shall be stored into an external DB.
>- Kamailio A is now u
On Mon, Oct 29, 2018 at 05:27:27PM +0100, Joan Salvatella wrote:
> On this setup we are facing 2 issues:
>- *Diversion headers access: *Currently, Kamailio only supports access
>to the last diversion header but since we are receiving traffic from Twilio
>(it sets the last Diversion head
On Mon, Oct 08, 2018 at 07:16:43AM -0400, Alex Balashov wrote:
> The SDP-bearing INVITE and response are simply passed along as-is by
> Kamailio, and it is the SDP which specifies where the media goes. So, if
> endpoint A calls through Kamailio proxy B to Asterisk server C via SIP,
> A and C will n
On Fri, Sep 28, 2018 at 11:47:12AM +0200, Pieter Muller wrote:
> If I insert my public IP under:
> # - start RTPProxy:
> #rtpproxy -l _your_public_ip_ -s udp:localhost:7722
> # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING
>
> Must I also modify UDP part below:
>
> #!if
On Fri, Sep 28, 2018 at 10:07:09AM +0200, Ivan Ribakov wrote:
> In a basic scenario of one2one call, I was able to make Kamailio to forward
> INVITE to callee over TCP by issuing a REGISTER request with
> ???transport=tcp??? parameter. Although it worked, as far as I understand
> that means all
On Thu, Sep 27, 2018 at 10:03:24AM +0200, Pieter Muller wrote:
> VoIP Supplier (IP:161.161.252.20{Private Net}) ? Connect via routing to
> Kamailio Server(155.155.16.2 eno2{private Net}),Kamailio
> Server(187.221.197.252 eno1{public facing})?connect to PBX(public IP set in
> dispatcher) ? connect t
On Wed, Sep 26, 2018 at 03:27:24PM +, Maarten Ureel wrote:
> 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:
On Thu, Sep 20, 2018 at 12:54:30AM -0300, volga...@networklab.ca wrote:
> User location lookup looks like can't handle long $rU like
> 10102-ce72256df4945bc472ed9c27a1037f46.
> It always return -3 404 not found.
This username is only 38 chars long, I have a username in a 5.1.4
environment with 3
On Thu, Sep 13, 2018 at 08:19:00AM +, Nathanael Eneroth wrote:
> UA1<--->KAM1<>KAM2<>UA2
> int1int1 int3 int3
> int2 int2
>
> I would like KAM1 to forward all SIP requests destined to int3 via
> KAM2 and vice versa. I am aware of the routing logic in
> 'kam
On Wed, Sep 05, 2018 at 06:36:20PM -0400, Alex Balashov wrote:
> Just grab it right before the consume_credentials() block, after all the
> challenge stuff.
consume_credentials() doesn't reset/clear $au, so anywhere after the
first if or after the route(that authenticates) will do.
signature.
On Mon, Sep 03, 2018 at 10:58:22AM +0100, David Villasmil wrote:
> Interesting approach! Though i don't think that will work for me. I've been
> looking into my requirements, and I'd need to do weight-based distribution
> instead of load-balance, also what i need is to add destinations to the
> di
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