Displaying 20 results from an estimated 25 matches for "trombon".
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trombone
2004 Sep 13
3
Astersk as AVAYA IVR
...all 6 PRI cards in the Index. (2 for termination to PSTN, 2 for outbound calls from Index to Asterisk and 2 for calls back from Asterisk to Index).
Is this feasible? Can anyone offer some tips/advice based on their experience? I've had a look around and I can't find anything relating to tromboning or anti-tromboning so I suppose each call will have to take the path of:
PSTN
|
Index
|
Asterisk
|
Index
|
Agent
Regards
Matt
2004 Aug 11
1
Blind Call Transfer using Sipura 3000 + asterisk
Hi List,
I hope this setup must be done by our astersik users..
I am using Sipura 3000 to receive PSTN calls and forward those calls to
asterisk for voice processing and after that, I am transferring call to
extension through FXS port on SPA 3000.
Currently, media of call is trombone through asterisk. i.e achieving blind
transfers on asterisk with SPA 3000.
Is it possible to stop trombone and send back media path on Sipura FXO to
FXS.
Current Call setup is
PSTN->SPA3000 FXO->Asterisk inbound--->Asterisk Outbound->SPA 3000
FXS->Analog Phone
I would like to ma...
2003 Aug 06
0
(no subject)
A new recording is now up on pan.zipcon.net and also gas.zipcon.net. Here is
the Readme:
The Virtuoso Trombonist
Dennis Smith plays with W.W.S.S Wind Ensemble Wm Cole, Conductor, and
with Martha Goldstein, organ. At ogg q=7.
<p>In the years befor world war 2, Sunday afternoons might be spent in the
park listening to the Municipal band. Selections 1 and 2 would be this kind
of "Pop" musi...
2004 Aug 10
1
SIP Transfers (Possibly reinvite)
Hey Folks,
Is it possible to transfer an incoming call back out without a "trombone"
effect.
For instance;
Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the
caller selects an option --> asterisk transfers the call to my cell phone
via broadvoice and removes itself from the equation so I end up with...
Caller --> Broadvoice --> Cell...
2004 Aug 12
0
Blind Call Transfer using Sipura 3000 + aste risk
...On Wed, 2004-08-11 at 12:00, Karun Chemudugunta wrote:
> I am using Sipura 3000 to receive PSTN calls and forward those calls to
> asterisk for voice processing and after that, I am transferring call to
> extension through FXS port on SPA 3000.
>
> Currently, media of call is trombone through asterisk. i.e achieving blind
> transfers on asterisk with SPA 3000.
>
> Is it possible to stop trombone and send back media path on Sipura FXO to
> FXS.
Basically you want to eliminate the back and forth traffic between the
Asterisk server, right? That'd be nice. :)
--...
2005 Aug 13
0
Re: Henning G. Schulzrinne quote on IAX2 from von magazine
...ote:
>On Sat, 2005-08-13 at 12:44 +0800, Steve Underwood wrote:
>
>> He doesn't seem to really understand the strengths and weaknesses of
>> IAX. IAX has drawbacks, but none of the problems he lists actually exist.
>
>OK, I'll bite ;-)
>
>How would IAX2 solve trombones?
>
>--
>Andreas Sikkema
Since this is a very vague accusation of protocol shortcoming, I'll
answer in a very vague way: IAX2 has the ability to "native bridge"
two endpoints together, even if the call was established by a third
party. Media is not "tromboned"...
2010 Mar 08
0
Is it possible to configure Asterisk so that it does the Q.SIG Path Replacement Feature ?
...B calls C (on the PABX) and does an attended transfer.
Is it possible to configure Asterisk so that it does the Q.SIG ?Path Replacement Feature? ?
The Q.SIG "Path Replacement Feature" requires the following:
After both legs of the call are setup and Asterisk has a successful "tromboned" or bridged call ...
Asterisk sees both 'B' channels (from trombone) are on same PRI/technology and initiates "Path Replacement" events
a. Asterisk sends "Transfer Complete" messages to both call legs
b. QSIG Switch sends "Purpose" message on one...
2017 Dec 07
0
Revolutions blog: November 2017 roundup
...html
An analysis of StackOverflow survey data ranks R and Python among the most-liked
and least-disliked languages:
http://blog.revolutionanalytics.com/2017/11/r-is-the-least-disliked-programming-language.html
And some general interest stories (not necessarily related to R):
* Siri transcribes a trombone player:
http://blog.revolutionanalytics.com/2017/11/because-its-friday-trombone.html
* A collection of short videos of interesting chemical reactions:
http://blog.revolutionanalytics.com/2017/11/because-its-friday-chemical-reactions.html
* An animation shows the impact of a rogue drone on Ga...
2020 Feb 20
0
anyone know of a list or wiki for GWC?
...thumps - No way I know of to do that. Audacity might be able to reduce the thump by using a high-pass filter set to roll off at 30 or 40 hz, but I don't think it can be taken completely out.
One of the really annoying things I remember about GWC was how it reacted to music featuring a bass trombone. The entire passage was detected as several million little clicks. When GWC got done, it sounded like the musician was playing through a fan. Totally unlistenable. Audacity's click filter can be adjusted to not do that.
--
Bill Gee
2004 Aug 04
2
Asterisk & ISDN-card
Hi!
If I install a CAPI-compatible ISDN-card in my server, will that:
a) enable me to connect that server to the public phone net
b) allow me to connect an ISDN phone to the server and use it as a SIP-phone
c) all of the above?
Regards,
Evert
2005 Feb 25
0
WG: AW: Transfer a call ? Am I looking for theflashcommand ?
...gt;> you are on and flashing it won't help.
>
> >> What you would need to do is get the other leg of the call to make the
> >> flash.
>
> >> Of course if you where on a PRI link, you could do "hairpinning", "ect"
> >> or "tromboning" and get the call taken back by the PSTN and transferred
> >> to the new number.
> >> --
> >> Steven Critchfield <critch@basesys.com>
--
Steven Critchfield <critch@basesys.com>
2007 Aug 29
0
Hangup detection and trombining
...detection issue
on 1 system (have a few others out there with TDM400's and no issue but this
one is causing a real headache)
The scenario is - system with TDM04B, a call comes in on a analogue line,
rings internally and then diverts to a mobile on a second analogue line, so
we in effect have a trombone happening where a call comes in on 1 analogue
and back out on another analogue.
Hangup detection seems to be working most of the time, but on a regular
basis does not (about once every 2 days or so). We cannot get hangup
supervision / polarity reversal or any other smart way of detecting a
hangup...
2001 Jun 28
0
Finale and Maestro fonts
...ld help, but it doesn't. I have
tried to add :
Alias0 "Maestro" ,"--maestro-", subst
to my wine.conf file, but it doesn't help.
Has anyone experienced / fixed / heard of the same problem? Finale is my
last tie to the evil side of my hard drive.
Joris
--
"Le trombone, c'est joli" - S. Gainsbourg
2001 Jul 27
0
Font / charset encoding problem?
...the fonts: I have tried demos of Finale, Noteworthy, Music Publisher,
Melody Assistant... with the same problem), someone might have run into a
similar issue with microsoft-symbol...
Any help would be GREATLY appreciated (actually, I could finally ditch my
Win partition!)
Joris
--
"Le trombone, c'est joli" - S. Gainsbourg
2007 Oct 18
0
Relaying calls to another SIP extension
...through Asterisk it 'rings' the
extension test at foo.bar.com as if it were a standard SIP phone.
That is stage 1.
Stage 2 is for the media server handling test at foo.bar.com to be able to
forward the call onto another SIP phone, allowing it to drop the call
completely and the call tromboning or bridging to happen in the Asterisk
PBX and not take up 'lines' on the media server. (Eventually both SIP
ends may become PSTN's).
I'm having trouble deducing how to do stage 1. In asteriskNow I don't
see a way to add a calling rule for this. I tried adding a service...
2005 Feb 24
1
Transfer a call ? Am I looking for the flash command ?
Hey Guys
Im trying to forward a call with asterisk to a regular phone.
Something like " I get a call on my regular phone, and he's trying to reach
some buddy of mine.. then I tell him "wait a sec" and push "Flash" and get a
other dialtone.. then I dial that other number then hangup the phone, so the
one that called will be connected to where I dialed it to"...
2003 Jun 28
0
SV: Newbie questions.....
...our
> main phone trunk to the public network. Can we connect an
> Asterisk PBX server with an E1 card to this? If so, could
> we then connect the Asterisk PBX to the callmanager?
> (Perhaps with another extension range).....and if so, how?
Yes, it's possible. I've done it to trombone calls through a
Operator system (Trio) and to interconnect with both Ericsson,
Nortel PBXs and iPBXs.
On CM you set it up as a trunk line, create a route map which
forward all calls to that specific E1 / T1 port on the 6608,
which are connected to the Asterisk Pri port.
On the Asterisk you do th...
2011 Jul 15
3
Redirecting call from one E1 to another?
I'd be grateful if anyone here could comment knowledgeably on an idea
that I have had, as to whether it should be possible or not.
Consider two Asterisk boxes, each with one or more E1s on EuroISDN.
Each box has a different telephone number that hunts across all its
E1 channels. In addition there is another number that hunts across
all the channels on all the boxes.
A call comes in to one of
2003 Nov 10
2
ISDN TBCT....
Greetings,
This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.
2020 Feb 20
3
anyone know of a list or wiki for GWC?
Hi!
totally OT...
Hoping there is a mailing list or wiki (or other help forum)
for GWC, but haven't found one yet.
I'm working on converting a bunch of my LPs to CDs, and am using
GWC (Gnome Wave Cleaner, or GTK Wave Cleaner) to clean up the noise.
It works great, but I can't figure out how to deal with certain types
of pressing flaws that create a thump every time around.
Anyone