Greg Thoen wrote:
4 Call comes in
dialplan calls specific lua script based on DID
lua uses luasql.mysql to get info from local mysql db
call is handled in lua using lua api
This is the option we chose in our set-up (although mysql is on a remote
server), currently we have over 25k DIDs
Anthony Minessale wrote:
i think you dial # instead of 0
There appears to be a bug, when I try this;
- A continues to hear moh
- B can't send any more DTMF or audio to C (but can still hear C)
- C can only send audio to B
I've posted the debug cli log on pastebin:
TTNC - Adnan Barakat wrote:
Is there a way to terminate the C leg when using att_xfer if the C leg
ends up being a voicemail?
Any ideas anyone??
Thanks
Adnan
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Hi,
Is there a way to terminate the C leg when using att_xfer if the C leg
ends up being a voicemail?
Thanks in advance
Adnan
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Mathieu Rene wrote:
It is possible that your inbound carrier applies some timeout rules.
Try the following before your bridge:
action application=ring_ready /
Not that I know of, I just tried with ring_ready, and it still doesn't work.
Thanks,
Adnan
TTNC - Adnan Barakat wrote:
Mathieu Rene wrote:
It is possible that your inbound carrier applies some timeout rules.
Try the following before your bridge:
action application=ring_ready /
Not that I know of, I just tried with ring_ready, and it still doesn't work.
Sorry guys, turns out it's
Hi,
I seem to have come across a strange problem; Basically I'm trying to
dial 3 destinations one after another, until the destination dialled is
answered, and I only want the destination to ring for 20 seconds.
If I do this from the console what I'm trying to achieve works fine;
originate
Hi
We have a FreeSWITCH server receiving calls from a provider, we process
the call (play greeting messages etc...) then pass the call out again to
an end destination via the same provider. But if the caller is
witholding their cli our provider send the call to us with the
sip_from_user
Michael Collins wrote:
I just did a test with this syntax and it worked for me. Please try it
and report back.
action application=bridge
data={ignore_early_media=true}[leg_timeout=20]sofia/internal/1...@1.2.3.4
mailto:1...@1.2.3.4|[leg_timeout=20]sofia/internal/4...@1.2.3.4
Anthony Minessale wrote:
Also is there any way to stop uuid_broadcast as I'd
need to stop it somehow if the destination picks up?
break uuid all
uuid_broadcast uuid phrase::saynumber,1 doesn't set the
'current_application_response' variable in the same way as
uuid_broadcast uuid
Chris Danielson wrote:
Excellent thanks, this is what I was looking for.
One last question if you don't mind; is there anyway to pull the caller
out of a fifo after a certain time either from api or by setting a
variable (eg. the destination didn't answer after sometime, so carry on
in
Anthony Minessale wrote:
what platform are you running on and what rev of the code?
SVN rev 13130 on Linux.
i suggest maybe you update your code or rebuild it clean.
Looking through my lua script, it seems timer_name=soft is causing the
problem, if I set this variable the quality becomes
Anthony Minessale wrote:
What scenario is this, what are you calling out to, there currently no
open issues mentioning anything about this?
I just updated and rebuilt FreeSWITCH to the latest trunk, everything
seems to be ok, uuid_broadcast works perfectly now.
Also is there any way to
Any ideas?
Is there no way uuid_displace can work on a non-bridged channel as it
appears to be the ideal solution?
TTNC - Adnan Barakat wrote:
Actually;
Just noticed that if I set fifo_music it doesn't work, whereas if I set
fifo_music to it works.
I just noticed this wasn't very clear, I
Just thinking - is there anyway to temporarily pause or silence the moh
in fifo via lua? as this would solve the problem.
Thanks
Adnan
TTNC - Adnan Barakat wrote:
Any ideas?
Is there no way uuid_displace can work on a non-bridged channel as it
appears to be the ideal solution?
TTNC
- Adnan Barakat wrote:
I just noticed this wasn't very clear, I meant fifo_chime_list only
works if fifo_music is blank.
Brian West
br...@freeswitch.org mailto:br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com
Hi All,
I'm trying to use uuid_displace to play a message to a caller who is in
the fifo queue, but uuid_displace doesn't play until the destination
answers.
I've tried uuid_broadcast, but the quality of the call becomes *really*
bad, then returns to normal quality after the file has finished
13130
Brian West wrote:
What SVN rev are you on?
/b
On Apr 23, 2009, at 9:44 AM, Adnan Barakat wrote:
Hi All,
I'm trying to use uuid_displace to play a message to a caller who is in
the fifo queue, but uuid_displace doesn't play until the destination
answers.
I've tried
Hi all,
I've just updated from r1 to the latest trunk (as I needed
mod_http), and group_confirm seems to have broken after the update. Now
the first 2-5 seconds of the file is played very quickly at poor
quality, then the end of the file plays fine.
Here is the relevant part of the
Brian West wrote:
Please update and try again.
Committed revision 10949.
Thanks Brian, works perfectly.
Adnan
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Adnan Barakat wrote:
Adnan Barakat wrote:
Got a strange problem when using originate_retries, if the destination
is busy/rejects the call there are 2 low pitched beeps heard, then FS
will try dialling again and the normal ring-tone is heard. (these 2
beeps are heard between very retry)
I
Hi Anthony,
Anthony Minessale wrote:
try latest trunk
Excellent, thanks a lot - works perfectly.
Adnan
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Adnan Barakat wrote:
Got a strange problem when using originate_retries, if the destination
is busy/rejects the call there are 2 low pitched beeps heard, then FS
will try dialling again and the normal ring-tone is heard. (these 2
beeps are heard between very retry)
I have removed
Adnan Barakat wrote:
Thanks Brian, I will update today, and will get back to you.
Adnan
Brian West wrote:
Please update to the latest SVN trunk and try again.
Fantastic, thanks a lot Brian for your help, it works perfectly now.
Adnan
Joseph Bajin wrote:
I assume you are meaning as you are connecting, you may want to play a
custom ringback or fake the ring.
Here's the page to do it:
http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
Or if you are using the originate command:
originate
Ashutosh wrote:
So, you want call from A to B, and make B listen to a file when he picks
up, right ?
You can originate the call to LegB , and make it drop to a context
which does the following
- first play the file
- Then bridge to LegA
Thanks Ashutosh, is this possible to do in an XML
Michael Jerris wrote:
Check out:
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
just setting the group_confirm_file without setting group_confirm_key
I think will do what you want.
Excellent thanks Mike, just tried that, however it won't work without
Michael Jerris wrote:
You are correct, You should be able to use:
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm
I tried this one, and in principle does exactly as needed however the playback
quality is really bad:
action application=set
Brian West wrote:
action application=set data=timer_name=soft/
Nope, still no luck, adding this setting didn't seem to make any difference.
Here is the actual dialplan I'm using on mod_xml_curl:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section
Hi All,
I'm looking for a way to play an audio file to the end destination before
bridging the call, eg. below is an example in Asterisk which I'm trying to
replicate in FS.
exten = _0ZX,1,Dial(SIP/outsip/${dest},30,rA(/srv/files/whisperfile))
Is this possible in FS?
Any feedback
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