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 modify
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
Just finished downloading the whole torrent and seeding now.
On Tue, Sep 29, 2009 at 3:04 PM, Dan White dwh...@olp.net 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
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
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 br...@freeswitch.org wrote:
I need the sip
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 thangappan...@gmail.comwrote:
Dear all,
I am in the process of implementing IVR server in Perl using
event outbound socket. I want to put a
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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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
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 from
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.
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
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 IP
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
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 :
action application=pre_answer/
action application=sleep data=2/
action
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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FreeSWITCH-users@lists.freeswitch.org
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:
This is unrelated,
use ring_ready instead of pre_answer
On Wed, Sep 30, 2009 at 11:26 AM, Keith Laaks kei...@voxtelecom.co.zawrote:
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
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
Dan Le dule.maill...@gmail.com 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
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