Wow! Alex and the kamailio list are at a perplexing loss. We wish you luck
with your future endeavors. We hope that your exclusive Acme contract does
not become some sort April fools joke to get you out of open source.
On Mar 31, 2013 6:33 PM, "Alex Balashov" wrote:
> For immediate release:
>
> A
This looks really awesome. Thanks for sharing
On Nov 19, 2012 4:50 PM, "Andrew Mortensen" wrote:
> I've been working on a Shared Call Appearances module for the past several
> months. It implements the Broadsoft SCA feature as laid out in Broadworks
> SIP Access Side Extensions Interface Specific
Is the database shared? If so maybe when they authenticate add a secure
token to the header that the second proxy can use for auth?
Just a suggestion not sure if its the answer your looking for or perhaps I
didn't understand the scenario well enough.
On Nov 19, 2012 7:53 AM, "Andreas Granig" wrot
any
>>>> of this B2BUAs in between the cards and Kamialio and get a limited capacity
>>>> application in front of Kamailio. !
>>>>
>>>> If there is any driver for kamailio where possibly use
>>>> media-proxy/rtpproxy for handling the PRI-Cha
Asterisk yate or free switch.
You need something as a gateway between PRI and sip. Kamailio does not
handle this conversion
On Oct 11, 2012 6:24 AM, "SamyGo" wrote:
> Hello,
>
> I've a scenario in which I've to deploy a couple Sangoma PRI cards with
> kamailio. What I wish is that I've some driv
Thanks Fred.
I think this was it.
I think there were some updates to the app_lua module since.
Do you know if this is still relevant?
On Sep 3, 2012 6:24 PM, "Fred Posner" wrote:
> Hi David,
>
> I believe this is the example you're looking for. It's on the Asipto KB
> site:
>
> http://kb.asipt
Sorry Daniel. I didn't see your message until I replied.
Understood.
On to 3.3...
On Jun 20, 2012 6:27 PM, "David J" wrote:
> Dave.
>
> Understandably. But my point was missed if you think that anyone here is
> trying to monopolize on the list please do u
Dave.
Understandably. But my point was missed if you think that anyone here is
trying to monopolize on the list please do understand that they usually
"contribute" to the project. Besides they are responsible for making
kamailio what is today. All so you can benefit. To say that there rates are
ou
You can use IP auth its simple and works.
On Mar 8, 2012 4:19 PM, "romon.zaman" wrote:
> hello room,
>
> i was trying to add sip provider(like. flowroute,vitelity) with user-pass
> authentication in kamailio to accept inbound calls.
>
> any help?
>
> thanks
>
> _
Yes set max-contacts in usrloc module
On Mar 2, 2012 5:45 AM, "Reda Aouad" wrote:
> Hi,
>
> Is there a way to ensure single-registration per user-agent for a user,
> which overwrites previous registration ?
> Or is there a way to limit the number of registrations per user, but
> overwriting the e
Hmm...we will see next year; :)
Seems like we have turned ClueCon into KamailioCon. Good work, all of you!
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Hello Daniel;
Will be there next week too; Going only for Thursday (Last Day);
Hope to have a chance to meet you.
Be in touch.
David.
On 8/2/11 5:15 PM, ifeanyi okoye wrote:
Hello Daniel,
My name is Ify and I'm a research voip engineer.I would be attending
the conference. Please can I re
Will you also be there the last day as well?
On Aug 2, 2011 6:15 PM, "Daniel-Constantin Mierla"
wrote:
> Hello Ify,
>
> On 8/2/11 11:15 PM, ifeanyi okoye wrote:
>> Hello Daniel,
>>
>> My name is Ify and I'm a research voip engineer.I would be attending
>> the conference. Please can I reach you via
I am using the kamailio default script on 3.1.3.
I was wondering what happens when I added an Alias in dbaliases?
For example if I add 18005551...@mydomain.com alias to 1...@mydomain.com
when an invite comes in; it works perfect I got a 200 back. (1001 Device
rings.)
If I add another alias t
Daniel;
Sorry for not being clear;
I understand the HTTP stuff;
I was asking if it is a simple as just calling the dialog bridge method
within the HTTP event route.
for example;
event_route[xhttp:request] {
dlg_bridge("sip:m...@myproxy.com", "sip:y...@yourproxy.com",
"sip:myproxy.com:5080"
Thanks Daniel;
What creates the Initial Dialog and REFER method
Just calling
dlg_bridge("sip:m...@myproxy.com", "sip:y...@yourproxy.com",
"sip:myproxy.com:5080");
When I get an HTTP event;
On 5/20/11 9:32 AM, Daniel-Constantin Mierla wrote:
Hello
I was wondering if I can make an HTTP request to Kamailio;
and then have kamailio do a lookup based on passed parameters
to connect callers.
I am trying to make a click-to-dial type application;
I was looking at the HTTP server inside kamailio;
It was interesting to me to try to use this as an a
Actually I might of made a mistake
I don't think the Voicemail table is used at all in that tutorial. Asterisk
only uses it. Kamailio does not use it.
On Apr 5, 2011 3:25 PM, "David J" wrote:
> Of course. Look at the config it should be very easy. Just replace with
the
> name
Of course. Look at the config it should be very easy. Just replace with the
name you want to use
On Apr 5, 2011 3:19 PM, "Lucas Alvarez" wrote:
> Hi, is it possible to change the name of the table voicemessages for
> voicemail profile in a kamailio-asterisk integration? I mean of the
kamailio
> s
For a second you really had us going.
Good Job
:)
On 4/1/11 4:54 AM, Olle E. Johansson wrote:
Friends,
After having spent many years working with the Asterisk SIP channel driver,
Kamailio and the SIPv2 protocol, I have finally realized that this is a dead
end. It's getting nowhere and it's w
5(20390) ERROR: [udp_server.c:586]: ERROR: udp_send:
sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file
descriptor(9)
5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed
I see this error when I try to restart kamailio after crash; I see the
cause of this problem is '
I realize that kamailio is not a b2bua;
But because we are using Asterisk in the path;
To extend the Asterisk Realtime Tutorial;
I was wondering if I could do something like this...
Kind of like how we use UAC to send a register to Asterisk;
Could we do the same and modify the method to use REF
I am trying to add support for call transfer in the Asterisk realtime
tutorial on Asipto;
I am not sure what I would have to do to get this feature working;
Perhaps I have to handle "refer" messages; but I am not sure how I send
that to Asterisk;
Any advice would be greatly appreciated.
_
if I do lookup()
What case does lookup return if entry exists but user not registered?
lookup("location");
switch ($retcode) {
case -1:
case -3:
sl_send_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Not Found");
exit;
};
should I
I am using the default script;
When I do a lookup for a user that is not registered; I get 404 back;
I added in my failure route to the list of codes 404; ie:
First on INVITE I have
t_on_failure("FAIL_ONE");
failure_route[FAIL_ONE] {
if (t_check_status("486|408|404")) {
sethostport("mydomain
If anyone can direct me how to resolve.
I load presence.so,pua.so,purple.so
When I run kamailio I see.
Dec 9 14:36:05 localhost /usr/local/sbin/kamailio[1605]: ERROR: purple
[purple.c:148]: can't import load_tm
Thanks.
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SIP Express Router (S
Hello List;
I was following along the Kamailio Asterisk Realtime Integration; I
modified the script to get it working on Kamailio v 3.1.
One area I had a question about was here...
We have the following block;
# Forward REGISTER to Asterisk
route[REGFWD] {
if(!is_method("REGISTER"))
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