Hi!
How can I configure voicemail so that I do not get the options such as
"record your message at the tone" and "mark this message as urgent"
Thanks!
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The bigger question is why was it finding that IP if it was wrong.
/b
On Jun 12, 2009, at 10:24 PM, John Wehle wrote:
Yep ... that's my SOP.
Looking at the internal.xml supplied with the new FS I see:
Once I commented out those entries everything worked fine.
I'm kind of surprised
Hi, I have a question about mod_voicemail. I would like to use it
independently of the dialplan and I was hoping to be able to add new
voicemail accounts on-the-fly, without having to update the FreeSWITCH
configuration files. But it seems to be forcing me to manually add each user
to the dialplan.
Hi all,
maybe someone can help me here.
For some compatibility issues I need to use an older version of t38modem which
doesn't support SIP registration.
Incoming faxes work well, but I don't know how to set this up for outgoing
faxes.
So, how can I setup this with FS for outgoing faxes without
Why would have to touch the dialplan? Just one regex to catch it
all.. and send it to voicemail. You still have to add users to the
directory for them to have a mailbox anyway.
/b
On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote:
Hi, I have a question about mod_voicemail. I would like to use
y to approach this?
> >
> > Thanks,
> > Adam
> >
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon! http://www.cluecon.com
>
>
>
>
> -
> Adam
>> >
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon! http://www.cluecon.com
>
>
>
>
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gt;>
>> Brian West
>> br...@freeswitch.org
>>
>> -- Meet us at ClueCon! http://www.cluecon.com
>>
>>
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>>
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hello
i am working for an austrian voip carrier.
for a few months i work with freeswitch and it is simply great.
it solves our needs in many places (high volume, flexible, stable).
the only thing i really miss is the avalibilty of php as a call control
language.
mod_php or mod_freehp are not comp
At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the
top under From Dialplan, it says:
Bridge the incoming call to extension 100 and 101. The '%' is used instead
of the @ to indicate that the endpoints are registered locally. Separate
multiple endpoints with a comma. The ${
Yes look at the default dialplan... you should note that its in the
default only.
/b
On Jun 13, 2009, at 2:42 PM, Lars Zeb wrote:
At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall,
near the top under From Dialplan, it says:
Bridge the incoming call to extension 100 and 101.
Perhaps you should look at controlling calls via the FreeSWITCH event socket
instead of from the dialplan. The nice thing about the event socket is that
your call control can happen on a separate machine. There is a PHP module
for the ESL (event socket library) and it would be relatively easy for y
I'm sorry, but I can't find ${sip_profile} defined in any document in the
conf directory.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Saturday, June 13, 2009 1:08 PM
To: freeswitch-users@lists.freeswitc
Its not there anymore... use_profile is. But its just a variable so
leaving it as is .. its prob. the best..
/b
On Jun 13, 2009, at 4:16 PM, Lars Zeb wrote:
I’m sorry, but I can’t find ${sip_profile} defined in any document
in the conf directory.
From: freeswitch-users-boun...@lists.free
Thanks. I would really like mod_voicemail to be database driven, instead of
by XML and cURL.
I noticed in the documentation that you can provide an ODBC handle in the
voicemail.conf.xml file, and so I tried it with MySQL.
It created the two tables, voicemail_msgs and voicemail_prefs properly, and
That does not provide for User configuration only voicemail metadata
storage... Again, at this time you would need to use XML CURL to do what you
are trying to do. There at some point in the future may be a way to grab
users via direct ODBC or other database driver however at this point its not
pos
Then you're free to write your own directory hook and plug it directly
into the database how ever you wish. Look how XML_CURL uses this
interface.
/b
On Jun 13, 2009, at 5:28 PM, Adam Wilt wrote:
> Thanks. I would really like mod_voicemail to be database driven,
> instead of by XML and cU
And please contribute things back when you do it.
Diego
On Sat, Jun 13, 2009 at 6:42 PM, Brian West wrote:
> Then you're free to write your own directory hook and plug it directly
> into the database how ever you wish. Look how XML_CURL uses this
> interface.
>
> /b
>
> On Jun 13, 2009, at 5:2
I couldn't agree more. We're working with a group that are developing a
massive PHP based music application. They are experts in PHP and MySQL but not
in VOIP/Telephony. By tuning an abstraction layer that uses PHP to communicate
with the FS event socket, allows them to work on the areas they
Ip Windows: 192.168.0.104
Ip Linux:192.168.0.105
My windows:
My sample on sip_profiles/external/dialus.xml
My linux:
My sample on sip_profiles/external/dialus2.xml
I have a num
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