Displaying 20 results from an estimated 900 matches similar to: "PlayTones problem"
2005 Feb 28
1
Zap channel calling back after hangup (due to polarity CID detection)
Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine.
I bought the card mainly to get caller ID to work properly in Sweden, and
that works just fine.
However, if the called or calling party hangs up after I hangup my SIP
channel, polarity CID detection kicks in and dials a couple of signals to my
incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have
tried
2005 Sep 30
1
Siemens TC35 GSM gateway
Hi all,
I have a TC35 and am keen to see if anyone has both voice and sms working from
Asterisk through this device? Google tells me that a few people have theorised
about it, I can't find anyone claiming to be doing it. What would be the best
way to put it into practice? Build a new channel for it?
Thanks
Andrew
2009 Sep 07
2
Echo and Playtones not working on SIP after upgrade
Hello list
I had the following echo-test extension on my Asterisk 1.2 setup.
exten => 1003,1,Wait(1)
exten => 1003,n,Playtones(!1050/1000)
exten => 1003,n,Wait(1)
exten => 1003,n,StopPlaytones
exten => 1003,n,Echo
exten => 1003,n,Hangup
After migrating my testing server to Asterisk 1.4, and a minor
extensions.conf update, everything works just fine. Except for the
Playtones
2019 Jan 31
2
Dailplan with playtones
Hello I use this dial paln:
[o2-in]
exten => o2,1,Answer
exten => o2,n,Playback(hello-world)
exten => o2,n,Ringing
exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt)
exten => o2,n,Playtones(425/1000,0/4000)
exten => o2,n,Wait(30)
exten => o2,n,Hangup()
All is fine. Hello world is Playback and I hear a ring tone.
If I remove the Playback hello-world. No ring
2014 Oct 30
1
PlayTones not working
I?m trying to use Playtones to have a tone played periodically throughout
phone calls. Unfortunately, I can?t seem to get PlayTones to work. I never
hear the audio tones.
Here is the output on the Asterisk console.
-- Executing [19525553312 at proxy-dial:2] PlayTones("SIP/testphone-00000032",
"1400/500,2000/5000") in new stack
[2014-10-30 14:28:31] WARNING[23154]:
2014 Oct 31
1
PlayTones while in call
I?ve gotten PlayTones to work, however it stops playing the tones as soon as
the call is answered. I would like to use PlayTones during the call because
I want to have a tone/beep played in the background while call recording is
going on.
Anyone know a way to get PlayTones to work while call is in progress?
Alternatively, does anyone have a suggestion for playing the tone/beep for
recorded
2005 May 10
2
skype channel
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I just noticed that the Skype API for linux seems to be available.
I've read before a number of posts where people were talking about
implementing a chan_skype with the skype API.
I wonder if there is any progress in that direction, and if anyone is
working on it.
/B
- --
* GPG-Key: http://evil.gnarf.org/mrbk.pgp
A: Because we read from top to
2004 Nov 25
1
Can't hear playtones?
Hello,
I would like the dialing party to know what happened to the call, since
asterisk doesn't relay a sip error back to the originating sip channel
(would be nice, a if (org_channel = sip && dst_channel = sip, relay error to
sip client) I want to set up audio feedback on the call status.
I've changed the county setting to NL in indications.conf and created this
test
2019 Jan 31
2
Dailplan with playtones
With softphone I mean linphone csipsimple or whatever.
How should a dialplan lokks like?
On 31.01.19 11:26, Antony Stone wrote:
> On Thursday 31 January 2019 at 10:59:01, basti wrote:
>
>> Hello I use this dial paln:
>>
>> [o2-in]
>> exten => o2,1,Answer
>> exten => o2,n,Playback(hello-world)
>> exten => o2,n,Ringing
>> exten =>
2009 May 27
1
Playtones Volume
I've researched my brains out on this, and can't find any answer. Is there
a way to adjust the level of the tones generated through the Playtones
command? I'm thinking that I may have been approaching this incorrectly by
targeting indications.conf since the tones are being called via the
Playtones application. My sense is that it's not possible due to the lack
of response from
2015 May 09
2
No application 'Playtones'
Hello Everyone,
We have most of the modules commented out. Can someone please let me
know which modules needed to be included for Playtones?
Kind Regards,
Nick.
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2005 Aug 26
1
bridging sip to capi, no playtones back to caller
I've the following setup :
sip phone -> ser (auth and routing) -> asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack
-- Executing Dial("SIP/2.7.184.61-08152880",
2004 Jun 08
0
TDM400P hangup / ringing detection problem
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Hi!.
I am having problems with getting asterisk to detect when someone hangs up.
I have a TDM400P with one FXO module connected to my telco, and also a
FXS-module connected to my phone.
The FXS-module detects hangups just fine, but I can't get the FXO to
detect them.
I am pretty sure i have disconnect supervision on my phoneline since
when I
2004 Jun 15
0
TDM400P FXO problems
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Hi!
I live in Sweden and I am having problems getting asterisk to properly
detect when a caller hangs up.
And yes, I DO have disconnect-supervision on my line.
Also asterisk sometimes misinterprets the disconnect-signal as another
incoming call. This usually happens if I hang up first and then when
the caller hangs up, asterisk treats it as a new
2005 Jan 13
1
Enabling/disabling zaptel echo-can from dialplan.
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Is it possible to enable/disable the zaptel echo-canceller
from the dialplan?
The reason I ask is that I want the echocanceller active on all calls except
when someone is sending a fax.
The simplest way would be to disable it on incoming calls to the fax
numbers and leaving it on on all other calls.
/B
- --
* GPG-Key: http://evil.gnarf.org/mrbk.pgp
2005 Sep 13
1
Integration between Asterisk and Siemens HiCom 150e over ISDN
Hi,
I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and
wondered if anyone is able to offer any advice.
In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk
box. e.g: HiCom user dials access code and can call Asterisk extension or
establish SIP call over Internet. Likewise, I'd like Asterisk to be able to
present a call to the Hicom, either
2004 Jun 08
2
Integration with a Siemens HiCom 150E / HiPath 3750
Hi * :-)
I found in the online WiKi docs some information on how to integrate
Asterisk with "old PBX"...
http://www.voip-info.org/wiki-Asterisk+legacy+integration
...but I couldn't find anything on integration with a Siemens HiCom
150E. Later on we'll migrate to a HiPath 3750 so information covering
this model would be nice too...
Do you know if any of the PBX listed
2006 Jan 18
1
bug in Authenticate application ?
I'm Japanese. Sorry,English is not so understood,Please let me question by
items.
In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of
Authenticate application's 'j' option.
exten => 123,1,Answer()
exten => 123,2,Authenticate(789,j)
exten => 123,3,Playback(pin-number-accepted)
exten => 123,4,SayDigits(111)
exten => 123,103,SayDigits(999)
In this
2007 Aug 21
1
Problems with overlap dial and Xorcom Astribank BRI
I have a strange problem with overlap dialing. I installed an asterisk
server between a Siemens HiCom PBX and our telephony provider.
Everything is working fine except some strange problems with the dialing
of the fax (connected to the HiCom PBX). It seems to me that if dialing
takes too long Asterisk just hangs up the channel without recognizing
that the fax machine is still dialing:
(Fax gets
2003 Jun 12
0
Playtones unexpected hangups
1) I'm working on a quick replacement for DISA, and I ran into the
following snag: When I specify "Playtones(dial)" I can only get
around 7 seconds of wait time before the dialtone stops, and the
context goes to the "h" extension. Is there a way around this fixed
timeout? The DigitTimeout setting doesn't seem to have any effect at
all on this hangup problem. I