Hi
> If you use a sofia profile with ODBC on multiple machines, if you register
> to any machine all of them will resolve the correct contact address when
> trying to make an outbound call. Presence, however such as subscriptions
> must still go to the original box because there is an open SIP d
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
momentarily.
The call scenario was a softphone (SJPhone) calling 3011 conferenc
I'm trying the third variant without much success. Here is what i do:
session.execute("ring_ready");
session.execute("set","ringback=%(2000,4000,440,480)");
session.execute("set", "transfer_ringback=%(2000,4000,440,480)");
The main problem i have is that there is no tone at all. No dial tone, an
On Mon, Jun 23, 2008 at 10:58 AM, Brian West <[EMAIL PROTECTED]> wrote:
> its 'park+5900' not just '5900'
>
> /b
I have this rockin' & it is working well. The issue is when the
caller is parked and then they hang up, the park indicator stays on
and park doesn't work right until you reboot the ph
http://dev.mysql.com/doc/refman/5.0/en/fulltext-fine-tuning.html
It could be something to do with the minimum length of a word when doing a
full text search.
Kind Regards,
Chris Mylonas
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On Mon, Jun 23, 2008 at 10:58 AM, Brian West <[EMAIL PROTECTED]> wrote:
> its 'park+5900' not just '5900'
>
> /b
>
That was it, I misunderstood what the config file was trying to tell
me! I ended up using 'park+1' - 'park+4'.
I will put a page on the Wiki to address it.
-Matt
_
check the packet flow in wireshark also
On Mon, Jun 23, 2008 at 2:06 PM, Brian West <[EMAIL PROTECTED]> wrote:
> By any chance is Asterisk involved in this at any point along the call
> path?
> /b
>
> On Jun 23, 2008, at 1:55 PM, UV wrote:
>
> I'm trying to send this and it doesn't seem to post
Sandeep,
In addition to the stuff Brian and Anthony asked you about, please do
these steps:
#1 - if you didn't make any modifications to your
conf/dialplan/default.xml file then delete it
#2 - in the fs source directory do "make current" which will update
everything
#3 - when the update is d
its 'park+5900' not just '5900'
/b
On Jun 23, 2008, at 3:52 PM, Matt Darnell wrote:
> On Fri, Jun 20, 2008 at 5:32 AM, Brian Snipes <[EMAIL PROTECTED]>
> wrote:
>> At least for Snom phones that is possible ( and used daily with FS
>> at my
>> office ). In the sample dialplan is a setting fo
On Fri, Jun 20, 2008 at 5:36 AM, Brian West <[EMAIL PROTECTED]> wrote:
>
> On Jun 20, 2008, at 2:11 AM, Matt Darnell wrote:
>
>> It must be hard to do, or everyone would be doing it.
>
> Its a matter of nobody can agree on how to do it.
>
>>
>>
>> If Freeswitch could offer the kind of functionality
On Fri, Jun 20, 2008 at 5:32 AM, Brian Snipes <[EMAIL PROTECTED]> wrote:
> At least for Snom phones that is possible ( and used daily with FS at my
> office ). In the sample dialplan is a setting for Snom's Park+Orbit button
> option. If you set the same Park+Orbit # on a button on all the phones
Gonzalo,
Agreed, the search term you were using should have worked. As I
mentioned in my previous email something is quite likely broken.
Hopefully we can get a little further in the investigation once we
know which release of MediaWiki is being used.
Jeremy
On 23-Jun-08, at 1:05 PM, Gonz
Nice work and thanks for making sure that the symptoms are not affecting
only me.
-MC
> -Original Message-
> From: Jeremy Kiffiak [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
> Kiffiak
> Sent: Monday, June 23, 2008 12:52 PM
> To: freeswitch-users@lists.freeswitch.org
> Cc: Michael Collins
Klaus,
I would like to thank you and anyone else that contributes to the
cause. It helps better the project.
/b
On Jun 23, 2008, at 3:19 PM, Klaus Teller wrote:
> Done. works fine with https.
>
> Klaus L'Imbecile.
>
> ---
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Done. works fine with https.
Klaus L'Imbecile.
Original-Nachricht
> Datum: Mon, 23 Jun 2008 15:12:25 -0500
> Von: Brian West <[EMAIL PROTECTED]>
> An: freeswitch-users@lists.freeswitch.org
> CC: [EMAIL PROTECTED]
> Betreff: Re: [Freeswitch-users] H323 support chip in
> try cha
On Mon, Jun 23, 2008 at 4:51 PM, Jeremy Kiffiak <[EMAIL PROTECTED]>
wrote:
> Michael,
>
> I have confirmed the issue with search myself and double-checked it
> against the search functionality of my own MediaWiki installation and
> that of Wikipedia. Something is definitely borked. It would appe
Oh btw you can go to freeswitch.org and click chipin on the right.
/b
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try changing it to https
/b
On Jun 23, 2008, at 3:06 PM, M.Emran wrote:
> me too.
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me too.
On Tue, Jun 24, 2008 at 1:56 AM, Seysan <[EMAIL PROTECTED]> wrote:
> I can not open the link!
>
>
> On Mon, Jun 23, 2008 at 1:16 PM, Brian West <[EMAIL PROTECTED]> wrote:
>
>> http://www.chipin.com/contribute/id/c648c4411bfb7861
>> Here is the URL to chip in for the H323 support in FreeSW
It redirects to paypal.. what does it do when you click it?
/b
On Jun 23, 2008, at 2:56 PM, Seysan wrote:
I can not open the link!
On Mon, Jun 23, 2008 at 1:16 PM, Brian West <[EMAIL PROTECTED]>
wrote:
http://www.chipin.com/contribute/id/c648c4411bfb7861
Here is the URL to chip in for th
I can not open the link!
On Mon, Jun 23, 2008 at 1:16 PM, Brian West <[EMAIL PROTECTED]> wrote:
> http://www.chipin.com/contribute/id/c648c4411bfb7861
> Here is the URL to chip in for the H323 support in FreeSWITCH.
>
> I'm a bit disappointed because so many of you out there were asking,
> beggi
Michael,
I have confirmed the issue with search myself and double-checked it
against the search functionality of my own MediaWiki installation and
that of Wikipedia. Something is definitely borked. It would appear
that searching by article title is successful but searching by
specific co
http://www.chipin.com/contribute/id/c648c4411bfb7861
Here is the URL to chip in for the H323 support in FreeSWITCH.
I'm a bit disappointed because so many of you out there were asking,
begging and really pleading for solid h323 support.
Please chip-in and lets make it happen.
Thanks,
Brian
By any chance is Asterisk involved in this at any point along the call
path?
/b
On Jun 23, 2008, at 1:55 PM, UV wrote:
I’m trying to send this and it doesn’t seem to post… here’s another
attempt:
After upgrading from latest stable version for windows (RC4) to
latest version, I’ve notice
I’m trying to send this and it doesn’t seem to post… here’s another attempt:
After upgrading from latest stable version for windows (RC4) to latest
version, I’ve noticed a significant loss of audio during a call.
It almost seems like there’s an enhanced VAD that triggers the audio on and
off –
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 from Zoiper (alex_gusak) to Zoiper (edward).
> ( Zoiper <-> SER <-> FreeSWITCH <-> SER <-> Zoiper )
> The f
Guys,
After Anton's post to the users list I did a search on "tdm" and "TDM"
on the wiki and got no results, not even page text results. This is bad
because the second sentence on the OpenZAP wiki page is:
The project implements a TDM abstraction library to provide an interface
to TDM hardwar
Anton,
My apologies for not responding to your first email. As Brian said, the
OpenZAP project is the piece that lets FS do TDM. In addition to the
OpenZAP section of the wiki (http://wiki.freeswitch.org/wiki/OpenZAP)
here's some quick info:
TDM cards from Digium, PIKA, and Sangoma have all been
Well it sure works better on 64bit OS and Hardware.
/b
On Jun 23, 2008, at 11:49 AM, Santiago Gimenez Ocano wrote:
This could be the problem. FreeSWITCH is designed for work in a
64bit OS? Do you think that the problem is because of the 64bit
hardware and i686 OS?
Also forgot to ask if thi
This could be the problem. FreeSWITCH is designed for work in a 64bit OS? Do
you think that the problem is because of the 64bit hardware and i686 OS?
Also forgot to ask if this was a 64bit OS?
/b
On Jun 20, 2008, at 5:27 PM, Santiago Gimenez Ocano wrote:FS1: Virtualized
Processor: Xe
As I wrote, we tested several clients.
One example of call from Zoiper (alex_gusak) to Zoiper (edward).
( Zoiper <-> SER <-> FreeSWITCH <-> SER <-> Zoiper )
The first client sends iLBC rtpmap: 98, mode = 30, FreeSWITCH sends to
second client iLBC rtpmap: 102, mode30.
Both the client did not hear
On Monday 23 June 2008 10:09:36 Anthony Minessale wrote:
> @([^:]*)
>
> no that is not me swearing cos of all the emails, that's the regex.
>
> again try the "info" app to see that most of the relevant uris are already
> cut up into variables for you.
Why not be IPv6-ready? Use
@(\[[^\]]*\]|[^:]*
@([^:]*)
no that is not me swearing cos of all the emails, that's the regex.
again try the "info" app to see that most of the relevant uris are already
cut up into variables for you.
On Mon, Jun 23, 2008 at 7:44 AM, Brian West <[EMAIL PROTECTED]> wrote:
> Please refer your local book store ab
route your inbound call to the "info" app. You will see the dozens of
things you can parse in the dialplan.
You can make a single extension that determines if it is a local domain or
not and formulate the dial string.
You have a lot of reading to do before you are authorized to comment on if
it's
What is the client on both ends of the call.
elaborate your diagram to show exactly what is on both ends of the call
including provider and it's name.
Many many places have a bug where they think that everyone should use the
same codec number as they do and do not deal with the reality that they
In order to do 2 things at once like that you need 2 threads.
Here are your options:
1) Use outbound mode of mod_event_socket to connect to a tcp port and
control the call from there.
2) Use the jsrun api call to launch another js in it's own thread and then
uuid_bridge the channels.
3) *Easiest*
Thanks for the input Mike.
The Wiki says that uuid_broadcast needs few arguments:
uuid_broadcast [aleg|bleg|both]
Any idea how these arguments should be passed to this application? I tried the
following without much success:
session.execute("uuid_broadcast",
"/usr/local/freeswitch/sounds/d
You have provided zero examples of what you're doing in terms of
FreeSWITCH dialplan can you include those so we can get a clear
picture of what you're doing?
Did you start with the default config?
/b
On Jun 23, 2008, at 1:52 AM, Sandeep Ammaladinna wrote:
Hi all,
I am very new to Fre
Example:
sofia/profile/[EMAIL PROTECTED]
/b
On Jun 23, 2008, at 5:36 AM, Sandeep Ammaladinna wrote:
Hi all,
Is it a mandatory requirement that for all outbound calls to an
external domain (domains that freeswitch is not responsible for), an
entry in the dial plan is required.
To be m
Please refer your local book store about info about regexp.
http://www.regular-expressions.info/
On Jun 23, 2008, at 4:57 AM, Sandeep Ammaladinna wrote:
Hi,
Can anybody help me out with a regular expression (in the context
of writing a dial plan) to extract the host part of a SIP URI.
wiki.freeswitch.org
Search for OpenZAP, Its a very extensive topic that has been talked
about and already works.
/b
On Jun 23, 2008, at 6:35 AM, Anton wrote:
> So, looks TDM is not interesting topic...
>
> On Saturday 21 June 2008 14:08, Anton wrote:
>> Hello!
>>
>> It's interesting to have a
So, looks TDM is not interesting topic...
On Saturday 21 June 2008 14:08, Anton wrote:
> Hello!
>
> It's interesting to have any info on how good would be FS
> in the TDM switching, including the proper timing and
> timing propagation among systems? Also many TDM hardware
> cards allows in-card fr
Removed from list.
On Jun 23, 2008, at 3:03 AM, "donald kingsley" <[EMAIL PROTECTED]>
wrote:
knowledge is wealth
donald has added you as a friend on BluBet.
Is donald your friend?
Please respond or donald may think you said no :(
Please do not reply directly to this email. This message
Hi all,
I am very new to Freeswitch and facing couple of issues related to
making outbound calls to external domains. Below is a detailed
explanation about the scenario I am trying and my setup.
1) I have freeswitch installed and running. I have configured a
domain within freeswitch by
Hi,
Can anybody help me out with a regular expression (in the context of
writing a dial plan) to extract the host part of a SIP URI.
Suppose I have a sip uri like "sip:[EMAIL PROTECTED]" or
[EMAIL PROTECTED] what is the procedure to extract the host part?
I know the procedure to extract th
Hi all,
Is it a mandatory requirement that for all outbound calls to an
external domain (domains that freeswitch is not responsible for), an
entry in the dial plan is required.
To be more precise, is it a pre condition that the name of the external
domain need to be know before making a call.
We have some problems with codec iLBC.
There is a scheme for passing the call:
++ ++
ClientA->||>||
| SER | | FreeSWITCH |
ClientB<-||<||
++
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