We wish to use iptel SIP server for our commercial product development. We plan
to develop SIP clients for Android, iOS, Windows phone and PC browser to have
voice and video calls.
Please let us know the licensing terms under which iptel SIP server is
available, especially in the context of us
Dear colleagues,We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE.Current setup is the following:PEER IP > KAMAILIO
Dear colleagues,We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE.Current setup is the following:PEER IP > KAMAILIO
Hi, I've been following this integration tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
and have a successful registration and I can even make calls through my
asterisk box.
However what is unusual to me is that every time a phone registers with
Kamailio, that
2015-07-15 21:59 GMT+03:00 Alex Balashov :
> As far as I know, t_on_branch() should execute for every branch, including
> the first one. Do you have any other evidence that it doesn't, i.e. a simple
> xlog statement?
I am unable to test this now in my local environment, but I believe we
have teste
On 07/15/2015 02:57 PM, Andrey Utkin wrote:
enable_double_rr is enabled by default in our config.
We use record_route_advertised_address() because other record_route
functions put internal IP to the header (the server is behind Amazon
firewall).
We call record_route_advertised_address() just one
2015-07-15 21:37 GMT+03:00 Alex Balashov :
> Andrey,
>
> Have you attempted to address this problem with the use of ordinary
> record_route() + the enable_double_rr option in the 'rr' module?
Alex.
enable_double_rr is enabled by default in our config.
We use record_route_advertised_address() beca
Andrey,
Have you attempted to address this problem with the use of ordinary
record_route() + the enable_double_rr option in the 'rr' module?
http://kamailio.org/docs/modules/4.3.x/modules/rr.html#idm10608
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North,
Hi!
We found that we need double Record-Route when we call from one SIP
transport to another (let's say, TCP -> TLS), because without
|transport=" hint in Record-Route, mobile Linphone app sends messages
(INVITE reply, ACK, BYE) via UDP, even if configured for another
transport.
record_route_adver
Roberto just call it on NATMANAGE as shown. I route everything now thru
that route..
You have to configure rtpengine to use internal external interfaces
route[NATMANAGE] {
#!ifdef WITH_NAT
# if (is_request()) {
# if(has_totag()) {
# if(check_route_param("nat=yes")) {
# setbflag(FLB_NATB)
Hi,
I am trying to extract and save ISUP from SIP-I packet for a later use. The
problem is that data is stored in binary format and needs decoding before
saving to a variable.
Can Kamailio perform any decoding so that later I can use append_body_part() to
append it back to a multipart message?
On 07/15/2015 08:44 AM, Alberto Sagredo wrote:
Hi Alberto,
can you also share part of the relevant place where you are calling that route?
Cheers,
Roberto Fichera.
> Hi Daniel
>
> Kamailio is for hard people and fun :)
>
> Thanks Visily i finnaly got it working with your tip. You were right ab
Hi Daniel,
We have of a set of CODEC which are all in public IP ( DMZ), we receive
a quantity of SIP calls (flooding) as such we plan to configure Kamailio
in Proxy SIP which works with a public IP ( DMZ).
I made the following modifications in the kamailio.cfg file:
* Related requests prese
You are welcome!
2015-07-15 15:42 GMT+02:00 Daniel Tryba :
> On Wednesday 15 July 2015 08:44:13 Alberto Sagredo wrote:
> > Kamailio is for hard people and fun :)
>
> No comment on this one
>
> > Thanks Visily i finnaly got it working with your tip. You were right
> about
> > internal extern
On Wednesday 15 July 2015 08:44:13 Alberto Sagredo wrote:
> Kamailio is for hard people and fun :)
No comment on this one
> Thanks Visily i finnaly got it working with your tip. You were right about
> internal external options instead direction=...
>
> Here its some code to someone could nee
We wish to use SIP-Router server for our commercial product development. We
plan to develop SIP clients for Android, iOS, Windows phone and PC browser to
have voice and video calls.
Please let us know the licensing terms under which SIP-Router server is
available, especially in the context of u
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