On Mon, Dec 29, 2014 at 5:03 PM, javier falbo
wrote:
> Hi,
>
> I would like some help from a kamailio expert.
> I have installed on a Debian7 64 kbytes VPS this tutorial:
> http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour,
>
> Problem is that there is no voice in a zrtp con
On Thu, Dec 18, 2014 at 9:35 AM, Thanh Truong
wrote:
> Hi Rob Moore,
>
> Yes, I have intended to use TLS in client side to verify with server side.
>
> I have tried to create cert files as :
> Quick Certificate Howto
> in http://kamailio.org/docs/modules/stable/modules/tls.html#tls.debugging
>
>
On Mon, Dec 1, 2014 at 12:17 PM, Javier Ricke wrote:
> Dec 1 16:15:22 serverjavier-desktop /usr/local/sbin/kamailio[7212]:
> ERROR: uac [uac.c:300]: 'append_fromtag' RR param is not enabled! - requir$
> Dec 1 16:15:22 serverjavier-desktop /usr/local/sbin/kamailio[7212]:
> ERROR: [sr_module.c:
http://www.kamailio.org/wiki/features/new-in-4.2.x
This page has links to the devel version of the modules docs. Would it make
more sense to link them to 4.2.x instead? The new-in-4.1.x page is the same
way, so I don't know if there's a specific reason you linked to devel.
Corey
_
On Thu, Oct 9, 2014 at 9:54 PM, logikanet wrote:
> Good evening!
>
> I've been trying to install Kamailio on a Centos 6.5 cloud , I'm able to
> compile and install everything, if I run : service kamailio start it
> starts kamailio and if I run service kamailio status I tells me that is
> ru
On Mon, Sep 22, 2014 at 10:46 AM, Tim Chubb
wrote:
>
> Hi
>
>
>
> I was wondering if the following scenario is possible and if it is how I
> would go about implementing it:
>
>
>
> I would like to be able read from a database a list of destinations
> associated with a given dialled number which a
On Wed, Sep 17, 2014 at 6:39 PM, aawaise wrote:
> How can we extract domain from the subscriber table to logs ??
> Its beyond my control at the moment to shift to kamailio-4.1. So have to
> persist with kamailio-3.1 ?
>
Look at the pseudovariable documentation. It sounds like $rd is what you
wan
The source tree includes an example of how to do this:
https://github.com/kamailio/kamailio/blob/master/examples/vm_proxy.cfg
After fr_inv_timer expires, Kamailio will generate a 408 and call your
failure_route, which is where you can redirect to your voicemail server.
On Wed, Sep 17, 2014 at 3:
On Thu, Sep 11, 2014 at 3:35 PM, Poekel wrote:
> Hello,
>
> I am totally new/blank at kamailio and would like learn more but do not
> know where to start
>
> I would like to build a voicemail platform with kamailio and Asterisk. (
> if i CAN use kamailio for this purpose)
>
Of critical i
On Sun, May 4, 2014 at 11:07 PM, aawaise wrote:
> Hello,
>
> I have configured TLS on kamailio server. Problem is that in kamailio.cfg,
> when I use command port =5061. Kamailio doesnot start and error given in
> logs is
> >> ERROR: tcp_init : bind(a, 0x9007f4, 16) on MyIP:5061 : Adress already
On Tue, Apr 29, 2014 at 12:06 AM, aawaise wrote:
> All certificates and keys have been made and updated.
> Can we make kamailio server to listen to two ports, i.e 5061 and 5062
> simultaneously ??
> Like in a scenario where some users connect to 5061 and some at 5062.
>
Yes. Simply add multiple
On Tue, Apr 15, 2014 at 9:47 AM, Slava Bendersky wrote:
> Hello Corey,
> Is there way publish address book through kamailio ?
>
>
As far as I know, that's a device-specific feature and not done through
SIP. For example, Polycom phones can read a directory.xml file to populate
their address book. S
On Mon, Apr 14, 2014 at 10:01 AM, Slava Bendersky wrote:
>
> Hello Everyone,
>
> Is possible do phone provisioning with kamailio ?
>
Kamailio itself does not do phone provisioning. It's not too hard to
integrate Kamailio with a system that does the provisioning though. Each
manufacturer has their
On Sat, Apr 5, 2014 at 2:03 AM, MrIhaveAnOpinionOnEverything <
melry...@gmail.com> wrote:
>
> Hi Everyone:
>
> I am a newbie in kamailio. Basically I am looking for a way to
> restrict the registration or login in kamailio per user to just one.
>
> The experience I want to setup is to unr
On Wed, Apr 2, 2014 at 12:36 PM, mark li wrote:
> Hi there.
> I've noticed that no matter what i do with my test phone (call voicemail,
> call another extension etc) I get error messages like the following in my
> log file:
>
> Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: ***
On Thu, Mar 27, 2014 at 2:31 PM, mark li wrote:
> Olle
> I've added some debug statements using the XLOG function like so:
>
> xlog("L_INFO", "my custom message");
>
> but none of my debug statements appear in syslog. I don't get any errors
> either when I restart kamailio.
> any suggestion?
> I
On Mon, Mar 24, 2014 at 1:28 PM, Alexandr Usov wrote:
> It is work for qualify - thanks.
>
> Looking for solution for external Asterisk subsribtion of presence states.
> Found the Pinan projec in the web, but it seems only Asterisk 1.4
> supported.
> I needs for Asterisk 11 or 12 version.
>
I no
On Tue, Mar 18, 2014 at 3:12 PM, Rene Montilva wrote:
> [server:192.168.1.1:5061]
> method = SSLv23
> verify_certificate = no
> require_certificate = no
> private_key = /etc/kamailio/key.pem
> certificate = /etc/kamailio/cert.pem
>
> [client:default]
> verify_certificate = no
> require_certificate
On Fri, Mar 14, 2014 at 5:13 PM, Pete Ashdown wrote:
> After starting to paste the kamailio.cfg snippets, I noticed I was using
> route(SIPOUT) instead of route(RELAY) for the aliases. Switching it to
> RELAY fixed it. What is the difference between the two? Does SIPOUT
> only work for devices
On Thu, Mar 13, 2014 at 8:26 AM, Pedro Niño wrote:
> The other (ugly) option, is to remove the auth from the phone, for the Sip
> Provisioning, but that would leave and open door to a reboot attack without
> auth needed from any IP. And I dont like that option.
>
This might not be as bad of an o
There's a typo in the Debian init script. Is this the correct place to
report packaging bugs?
--- /etc/init.d/kamailio2014-03-06 13:42:23.0 -0700
+++ /tmp/kamailio2014-03-07 08:41:05.0 -0700
@@ -52,7 +52,7 @@
log_failure_msg "Not starting $DESC: invalid configurati
On Fri, Feb 21, 2014 at 12:18 AM, Owais ul Haq wrote:
> Hello,
>
> I have deployed Kamailio-3.1 on a fedora machine. And my database is
> placed on another windows 2008 server situated on the local network.
> Problem is when I run kamailio. I get the following error from logs.
>
> [cfg.y:3416] :
On Fri, Feb 14, 2014 at 7:35 PM, wrote:
> Testing in the same box for now with the goal to at least get it working
> within one machine.
> Do you mean that TLS will not work with the cert/key shipped with kamailio?
>
I've never tried. A default key would not be very secure, but if you have a
va
On Thu, Feb 13, 2014 at 4:35 PM, wrote:
> Corey:
>
> listen=tls:127.0.0.1:5066
>
Make sure in your config that 127.0.0.1 is replaced by your real IP address,
or 0.0.0.0, otherwise it will only listen on the local host and won't be
open over the network.
> port=5066
>
This is unnecessary.
>
On Thu, Feb 13, 2014 at 9:52 PM, Owais ul Haq wrote:
> Hello,
>
> I have installed kamailio server. Now I am trying to use SSL for my
> server - client connections.
> I have made certificates succesfully. Now when my client connects, I get
> the following error.
>
> [tcp_read.c:882] : ERROR : TL
On Wed, Feb 12, 2014 at 8:11 PM, wrote:
> Kamailio running in Fedora 19 box. The client (Jitsi) can only connect
> over TCP.
> If connecting over UDP, the clients connect, but cannot send/receive
> messages.
> Cannot connect over TLS at all. Ultimately I will need to run with TLS
> over 5065 port
On Mon, Feb 10, 2014 at 5:03 AM, Daniel Goepp wrote:
> Try running this command to see what application is using that port:
>
> netstat -ulnp
>
>
TLS is a TCP port.
netstat -lntp |grep 5061
Corey
___
SIP Express Router (SER) and Kamailio (OpenSER) - s
On Thu, Feb 6, 2014 at 3:26 AM, jaflong jaflong wrote:
>
>
> This is my tls.cfg for server
>
> [server:default]
> method = TLSv1
> verify_certificate = no
> require_certificate = no
> private_key = /etc/asterisk/certs/proxy.key
> certificate = /etc/asterisk/certs/proxy.crt
>
>
> As far as I under
On Tue, Nov 26, 2013 at 7:58 AM, Joli Martinez wrote:
> Hello,
>
> I am getting a lot of hits on port 5060. I would like all registered users
> to be sent to the correct location (currently working), but all unwanted
> users allow them to connect and any number they dial it will alway play a wa
On Thu, Nov 21, 2013 at 5:49 AM, Grant Bagdasarian wrote:
> Is it possible to have the sipcapture module write the duplicated messages
> to multiple tables?
Not that I can see. Could you use MySQL replication?
Corey
___
SIP Express Router (SER) and Ka
On Wed, Nov 6, 2013 at 12:03 PM, Daniel-Constantin Mierla
wrote:
> On 11/6/13 2:58 PM, Alex Balashov wrote:
>> 2. Is there any harm in calling unforce_rtp_proxy() for Call-IDs rtpproxy
>> doesn't know about? is there a 'better' best practice for handling CANCELs
>> where it is unknown whether rtp
On 01/30/2013 07:36 AM, Alex Balashov wrote:
> On 01/30/2013 09:31 AM, Julia wrote:
>> We need some number manipulation for outgoing calls to PSTN GW.
>> The same manipulations must be in "ru" and "tu", because our PSTN GW adds
>> redirection when tu ≠ ru.
>> When we used INVITE re-parsing for CANC
On 06/26/2012 08:36 AM, Gertjan Wolzak wrote:
> Which timer do I need to use to get a failure route to reroute the call
> to a pstn gateway?
>
>
>
> I have tried the fr_timer which works, I set it to 5000, downside is
> that when I am in wifi range, I only have 5 seconds to answer the call.
>
On 10/06/2010 11:47 AM, Shrouk Khan wrote:
> hi,
> i have been looking for a way to authenticate users based on the IP from
> they come from .
There are a number of ways you could do this. They way I did it by using
a group named ipauth. If the user is a member of the group, compare the
request IP
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