Hi All,
I gave this a shot. And it works. But I am facing a unique issue.
What I have done is created a view called sip_registrations for FS from
the schema for location table in OpenSER. (For details please refer my
earlier mails)
Now when I register a user, the registration is successfully d
Correct. The conference is unusable. When 2 people in the conference they
can not hear each other or can hear each other intermittently.
I’ve played with the energy level and flags settings but without any luck.
The pcap I’ve done is on the client. When doing pcap on the FS box I get the
sa
the part that builds that page is in mod_shout in the web_callback
function.
/b
On Jun 24, 2008, at 2:10 PM, e schmidbauer wrote:
does anyone know where i can find the source files for this page:
http://freeswitchip:8080/api/telecast/index
On Tue, Jun 24, 2008 at 3:00 PM, Michael Collins <[
so the conference is never working you are saying?
When 2 people are in the conference can they hear each other?
are you doing pcap on the FS box or on the client side?
There is energy detection in the conference that you could try turning down
in the conference profile.
try this setting to mak
does anyone know where i can find the source files for this page:
http://freeswitchip:8080/api/telecast/index
On Tue, Jun 24, 2008 at 3:00 PM, Michael Collins <[EMAIL PROTECTED]>
wrote:
> Check mod_shout.c
>
>
> --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PR
Check mod_shout.c
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of e
schmidbauer
Sent: Tuesday, June 24, 2008 11:37 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout.c:497 write_stream_thread()
MP3encode error
is pretty much the same. I can’t tell if 90% or 70% of the traffic is
lost but something definitely makes the FS stops sending traffic and start
again (as if it was VAD).
The client sends RTP as usual. No change there. The loss is only from the FS
server side.
Problem persists when adding ca
is there somewhere i can find the source code for that page?
On Wed, Jun 18, 2008 at 6:04 PM, Brian West <[EMAIL PROTECTED]> wrote:
> If you have mod_shout and mod_xml_rpc loaded here is a little
> something nobody knows but I do expect someone to put on the wiki..
> please please please
>
>
> Go
FYI,
Brian recently committed a patch that adds a new extension to the
dialplan: 9992, aka "show_info".
It simply does an info app then hangs up. We added it so that you have
a quick and easy means of doing simple testing. For example, if you
have a SIP phone registered you can just dial
so are you also saying that the client is not sending RTP either?
did you try adding subsequent callers to the conference?
is this on more than one box?
is this only on windows?
On Tue, Jun 24, 2008 at 11:37 AM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> what happens when you call ?
>
>
what happens when you call ?
2008/6/23 UV <[EMAIL PROTECTED]>:
> I've checked the packet flow in wireshark and it seems like the FS is
> sending RTP for about 2 seconds (the "alone announcement") and then stops
> sending RTP. From time to time few packets arrive and you can hear the MoH
> m
Anton,
what do you mean by 20 ports? (20 B channels, 20 pri spans? 20 cards)?
And why is 20 a limiting factor?
Just curious (as it wasn't clear to me).
Thanks
Tony Knight
On Tue, Jun 24, 2008 at 6:05 AM, Anton <[EMAIL PROTECTED]> wrote:
> Michael,
> Thanks for reply.
>
> Yes, I do use SANGO
This looks like they are not understanding that our re invite is for
the same call
Mike
On Jun 24, 2008, at 6:41 AM, Ivan C Myrvold <[EMAIL PROTECTED]> wrote:
I got email from support today, so hopefully they will find out why
this happens, although they asked me to try out some other ports
It works! Thank you for fixing this problem.
2008/6/23 Anthony Minessale <[EMAIL PROTECTED]>:
> please try latest trunk, I may have figured it out.
>
>
> On Mon, Jun 23, 2008 at 10:07 AM, Alex Gusak <[EMAIL PROTECTED]> wrote:
>>
>> As I wrote, we tested several clients.
>> One example of call fro
I got email from support today, so hopefully they will find out why
this happens, although they asked me to try out some other ports,
which I did and the same happened.
Ivan
Den 24. juni. 2008 kl. 11:10 skrev Ivan C Myrvold:
I have a softphone, iSoftPhone, I am trying out, but when register
Michael,
Thanks for reply.
Yes, I do use SANGOMA TDM cards in my systems - have several
types of their TDM and analogue hardware (A200,
A102,A104,A108,108D)
Right now my most capable system runs latest asterisk with
20 ports TDM, SS7 and PRI (chan_ss7), though call volume is
not too big, and
I have a softphone, iSoftPhone, I am trying out, but when registered
to FreeSwitch, it behaves strangely. That is probably the softphone's
fault, but I thought I would report it here, so I could provide the
developer of this phone with some more intelligent response.
I have iSoftPhone on th
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