Hello,
I must say this problem with Asterisk versions before 11.11 remains a
mystery, after upgrading to Asterisk 11.11 and a while of setting up
Asterisk realtime fields and also upgrading to sip.js I managed to get
calls flowing between websocket and Zoiper clients.
In case someone is having si
Hi,
Thanks for your efforts, now after lots of hours trying different ways and
working through my config, I'm baffled. Somehow I think I must have done
something wrong when combining different tutorials (like use of dispatcher,
realtime integration and websocket clients). Something I noticed was t
On 24/07/14 09:27 AM, Olli Heiskanen wrote:
That's odd... I pulled a new version from git master 4 days ago, and
copied the compiled rtpengine to /usr/sbin, which is running. (although
might help verifying the version if command rtpengine --version gave
actual output instead of 'undefined') :)
That's odd... I pulled a new version from git master 4 days ago, and copied
the compiled rtpengine to /usr/sbin, which is running. (although might help
verifying the version if command rtpengine --version gave actual output
instead of 'undefined') :)
Any chance my environment might cause something
On 07/23/14 11:01, Olli Heiskanen wrote:
>
> Thanks,
>
> I think here's all of the call from before the called party answers:
...
I can't seem to reproduce this, when I run through the same sequence,
the answer comes out as a=setup:active as it should. Are you sure you're
using the latest vers
Thanks,
I think here's all of the call from before the called party answers:
Jul 22 19:36:31 u363id562 rtpengine[16930]: Got valid command from
127.0.0.1:39090: offer - { "sdp": "v=0#015#012o=- 4878041229845783313 2 IN
IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE
audio#015#012a=ms
On 07/23/14 05:03, Olli Heiskanen wrote:
>
> Hi,
>
> Thanks very much for this, that solved the double-m-line issue. Now I'm
> calling rtpengine_offer in a branch route.
>
> One issue still remains; the call still gets connected to the called
> zoiper client, but it gets hung up right away. I tr
Hi,
Thanks very much for this, that solved the double-m-line issue. Now I'm
calling rtpengine_offer in a branch route.
One issue still remains; the call still gets connected to the called zoiper
client, but it gets hung up right away. I traced this to be caused by a BYE
message from Kamailio, whi
On 20/07/14 01:15 PM, Olli Heiskanen wrote:
Hi,
...
There may be something off in my Asterisk configs since it's Asterisk
that responds 488, but see how Kamailio responds, SDP contains 2 similar
m= lines. Is there something I might be doing wrong in configuring
rtpengine? The INVITE going to th
Hi,
After a long break while concentrating on other parts of my system I'll get
back to this problem. My system has changed a bit since the last setup,
although it's mostly the same: CentOS 6.5, Kamailio on 1.1.1.1:5060 and
newly added Asterisk on 1.1.1.1:5070. I use realtime integration, which
se
On 04/12/14 09:31, Olli Heiskanen wrote:
> Hello,
>
> I'm probably still doing something wrong, I still get 488 from the
> grandstream. Also zoiper refuses the call with 415 Unsupported Media Type.
>
> According to the module description I tried to change my config to this:
> Btw, thanks for enab
Hello,
I'm probably still doing something wrong, I still get 488 from the
grandstream. Also zoiper refuses the call with 415 Unsupported Media Type.
According to the module description I tried to change my config to this:
Btw, thanks for enabling verbose flags, those are more readable when
workin
On 04/10/14 09:26, Olli Heiskanen wrote:
>
> Hello,
>
> After some tests, I'm still having some strange results.
>
> When calling from ws client to grandstream, I get the below output to
> /var/log/messages.
> In a sip trace after 488 there are only INVITEs from kamailio server to
> grandstream
Hello,
After some tests, I'm still having some strange results.
When calling from ws client to grandstream, I get the below output to
/var/log/messages.
In a sip trace after 488 there are only INVITEs from kamailio server to
grandstream but no responses come back to kamailio server.
I haven't ch
Indeed, which works for simple demos and fits on a single slide - the whole
purpose of that presentation. If someone is building a production system
they really need to understand the various use-cases they will see and
write their Kamailio configuration properly.
Regards,
Peter
On 6 April 2
Hi,
Thanks, it compiled nicely, I'll continue with more testing tomorrow.
- Olli
2014-04-08 15:36 GMT+03:00 Richard Fuchs :
> On 04/08/14 03:00, Olli Heiskanen wrote:
> > Hello,
> >
> > Thanks Juha, that will be a good thing to investigate more when I get my
> > simple unrealistic scenario wor
On 04/08/14 03:00, Olli Heiskanen wrote:
> Hello,
>
> Thanks Juha, that will be a good thing to investigate more when I get my
> simple unrealistic scenario working. :)
>
>
> I tried compiling rtpengine on Centos 6.5, I wonder do I need to change
> the Makefile somehow for CentOs? Remove Debian
Hello,
Thanks Juha, that will be a good thing to investigate more when I get my
simple unrealistic scenario working. :)
I tried compiling rtpengine on Centos 6.5, I wonder do I need to change the
Makefile somehow for CentOs? Remove Debian specific flags like mentioned in
the github page? Below
Olli Heiskanen writes:
> Thanks, I'll look into the rtpengine, had a busy weekend but next week I'll
> have better time.
what comes to peter's slideshare failure_route example, i think it only
works in very simple unrealistic scenario when there is no forking or
serial routing. also, its nathelp
Hello,
Thanks, I'll look into the rtpengine, had a busy weekend but next week I'll
have better time.
The function seems like a good idea. I'd definetely rather use that if/when
it's available.
cheers,
Olli
2014-04-04 19:12 GMT+03:00 Juha Heinanen :
> Olli Heiskanen writes:
>
> > if ( sdp_get_
Olli Heiskanen writes:
> if ( sdp_get_line_startswith("$avp(mline)", "m=") ) {
> if ($avp(mline) =~ "SAVPF") {
in order to simplify the above, how about introducing a new function
sdp_with_transport_like(transport)? that function would return 1 if
string of the param is included in the transport
On 04/03/14 15:32, Olli Heiskanen wrote:
> Hello,
>
> Thanks, I'll give that a try and post back. I guess I install and run it
> just like mediaproxy-ng?
Yeah, pretty much. Lots of internal changes, but externally the biggest
change is the name.
> I'll also try different sip clients like zoiper
Hello,
Thanks, I'll give that a try and post back. I guess I install and run it
just like mediaproxy-ng?
I'll also try different sip clients like zoiper etc.
One thing that occurred to me based on the fact that the sdp is faulty, as
I did this test from the slides here:
http://www.slideshare.net
Hi,
This seems to be caused by an additional media stream (second m= line)
appearing in the answer SDP, which is invalid according to RFC 3264.
I'd like to invite you to try the upcoming new version of mediaproxy-ng
instead, which has been renamed to rtpengine:
https://github.com/sipwise/rtpengin
Hey,
Unfortunately your packet dumps are truncated and don't show the
complete SDP bodies. It would also be interesting to see which options
and parameters are passed to mediaproxy-ng when processing the SDP. You
would find this info in the log produced by mediaproxy-ng, which should
also include
Hello,
I've been experimenting with Kamailio with ws and sip clients and could
need a hand in getting a call between those two to work.
I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS
6.5 and a mediaproxy-ng running. I have clients wscli...@testers.com and
gscli...@teste
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