Dan Le wrote:
> We're running into a problem with the minimum file size when recording using
> uuid_record. It seems if the audio is too short it deletes the audio file.
> Is there a way to override that?
Yes. It was discussed on the list recently. I suggest searching the list
archives. Someone m
We're running into a problem with the minimum file size when recording using
uuid_record. It seems if the audio is too short it deletes the audio file.
Is there a way to override that?
We're running the FS 1.0.4 tag, on a windows xp 64bit machine. We start
uuid_record from a lua script, recording a
This is unrelated,
use ring_ready instead of pre_answer
On Wed, Sep 30, 2009 at 11:26 AM, Keith Laaks wrote:
> Hi,
>
> I am testing a trunk version from the weekend and have moved configs
> over from a box pre 1.04. Looks like Rev: 15011, but the box was built
> by someone else - so not 100% cer
Hello,
At r15025 the ringback tone handling has changed.
The old method:
When ringback was set, early media was ignored, meaning that if you got
183 with SDP, ringback was generated locally.
The new way:
When ringback is set, the ringback tone is generated only when there is
no early media (th
and (if using custom_sql + the right query) you can now have the
channelvar lcr_user_rate populated with your end-user-rate tables
which can then be used by mod_nibblebill or other billing engine.
Just committed and wiki updated.
On Wed, Sep 30, 2009 at 11:30 AM, William Suffill
wrote:
> You cou
Thanks.
That did the trick and now FS sends the correct address.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, September 30, 2009 7:11 PM
To: freeswitch-users@lists.fr
You could also use mod_lcr to query the cost to aply to the nibblebill as well.
ODBC keeps the connection to MySQL open as needed and reconnects if need be.
-- W
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Hi,
I am testing a trunk version from the weekend and have moved configs
over from a box pre 1.04. Looks like Rev: 15011, but the box was built
by someone else - so not 100% certain of exact number.
With :
I am not getting ringback after the 'pre_answer'.
Was testing with G722 and G729 on th
I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone. Does anyone know why?
Best Regards,
Jerry
On Wed, Sep 30, Andrey Nepomnyaschih wrote:
> No luck.
>
> These networks (192.168.50/24 and 192.168.60/24) are now connected through
> another transit network (10.15.118.252/30) without using any firewall or
> tunneling software. But FS still asks that endpoint to send RTP media to
> external
Its because your local-network-acl doesn't include that other network
please correct that and it'll work.
/b
On Sep 30, 2009, at 10:57 AM, Andrey Nepomnyaschih wrote:
> No luck.
>
> These networks (192.168.50/24 and 192.168.60/24) are now connected
> through another transit network (10.15.11
No luck.
These networks (192.168.50/24 and 192.168.60/24) are now connected through
another transit network (10.15.118.252/30) without using any firewall or
tunneling software. But FS still asks that endpoint to send RTP media to
external IP address and I'm really lost with this problem.
-
Nm I see it.. not awake this morning :P
/b
On Sep 30, 2009, at 7:06 AM, Fernando Testa wrote:
> Brian,
>
> Thanks for the reply. The SIP trace is mixed with the log (+ sofia
> loglevel all 9) on the pastebin I mention on the previous email
> (http://pastebin.freeswitch.org/10517 ). That log is f
I don't see a challenge in your 407 so how can we answer properly
against the far end if they don't challenge us?
/b
On Sep 30, 2009, at 7:06 AM, Fernando Testa wrote:
> Brian,
>
> Thanks for the reply. The SIP trace is mixed with the log (+ sofia
> loglevel all 9) on the pastebin I mention on
Park it.. then send it a command to play music.
/b
On Sep 30, 2009, at 7:47 AM, Thangappan.M wrote:
>
> Then what is the right way to put a call on hold and play the music.
> Can you please explain?
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Then what is the right way to put a call on hold and play the music. Can you
please explain?
On Wed, Sep 30, 2009 at 10:57 AM, Thangappan.M wrote:
> Dear all,
>
>I am in the process of implementing IVR server in Perl using
> event outbound socket. I want to put a call in music on ho
Brian,
Thanks for the reply. The SIP trace is mixed with the log (+ sofia
loglevel all 9) on the pastebin I mention on the previous email
(http://pastebin.freeswitch.org/10517 ). That log is from FS 1.0.4.
On Tue, Sep 29, 2009 at 10:13 PM, Brian West wrote:
> I need the sip trace.
>
> /b
>
> On
Hi,
Thanks for reply. Actually I'm currently running asterisk and for billing
I'm using MySQL database. In the database i've listed various destination
against their different rates. When call is made a script which queries
MySQL for billing. My question is, how can I do this in FS using mod
nibb
Just finished downloading the whole torrent and seeding now.
On Tue, Sep 29, 2009 at 3:04 PM, Dan White wrote:
> I've downloaded 0Kb in 56 minutes. Anyone mind setting up a seeder? I'm on
> a pretty fast connection.
>
> While waiting, I found the following discussions:
>
> I found the following
This API is not for that. It actually sends the hold notification to
the endpoint. It's not for starting or stopping HOLD music. You
could use it as such if you had TWO freeswitch boxes...
CALL -> FS -> FS (uuid_hold would send a hold indication to the first
FS box causing it to play mu
Yes set max_forwards to what ever you want.
Hint its a variable.
/b
On Sep 30, 2009, at 2:26 AM, Helmut Kuper wrote:
> hm, well that means that FS may has to work on a messages around 70
> times before it is destroyed. 70 times of signaling work in maximum
> performance I guess. Any way to modi
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hm, well that means that FS may has to work on a messages around 70
times before it is destroyed. 70 times of signaling work in maximum
performance I guess. Any way to modify default max forward header?
On 29.09.2009 18:56, Michael Jerris wrote:
> we
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