Displaying 20 results from an estimated 100 matches similar to: "Playtones not passing sound to incoming SIP connection"
2005 Sep 13
1
sometimes dtmf passed, sometimes not (cisco 7960 SIP)
Hi list, I'm hoping that I'm being stupid, and someone can tell me
what's going on, but for the life of me I can't figure it out. (it's
been a long day, and I'm now in the last 3 weeks of organising my
wedding, so I hope this makes sense ;) )
When at my desk, accessing (for example) my voicemail, the dtmf tones
are passed perfectly, I can enter password, change
2004 Dec 23
2
Incoming calls from Sipgate go through the wrong peer
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it seems to pick the
wrong peer from sip.conf which sends the call into the wrong context and
it fails because there is no extension in that context to match the
register.
Using the config's below, if I dial the DiD on account 2222222, it works
fine - picks peer 2222222
2006 Feb 25
2
sipgate.de question
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n)
I looked at the sip debug stuff, and all I
2005 Feb 01
0
Limiting no. of calls on one channel
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup
-Matthew
----- Original Message -----
From: "Stefan Gofferje" <stefan@gofferje.homelinux.org>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Monday, January 31, 2005 6:43 PM
Subject: [Asterisk-Users] Limiting no. of calls on one channel
2008 Mar 07
3
Silencing VoiceMail() app in * 1.4.10
Hi there,
Googling through the archives it looks like I'm the ferst person to want
this...
My aim is to set up a voicemail application with a custom greeting before
*AND AFTER* the punter has left the message.
Right now the relevant section of my dialplan is like this:
exten => 2,1,Playback(/media/asterisk/answerphone-en)
exten => 2,n,VoiceMail(2000,s)
exten =>
2004 Sep 15
2
Results of 13 month study on reducing telemarketing calls
Hello--
I've been playing with the privacy options on my home/home-office system
since August last year, and have some results, gleaned from my CDR
records, which over the last 13 months, number a total of 8672, which
includes incoming, as well as outgoing calls.
Before I start spitting out numbers, let me note that with the current
setup, I haven't had to tell a single telemarketer
2003 Jun 03
0
Sound: Recording overrun
Hello All,
I've just been made aware of Asterisk and have installed it. Things seem
to be working: I can dial from the console the internal numbers and hear
the answerphone messages and do other interesting stuff.
however when I hangup the call I see a continuous stream of messages
saying "Sound: Recording overrun" appear on ALL my linux consoles and I
can't see what I'm
2005 Sep 02
0
monitoring VM via speaker and grabbing connection
This is an old problem, but I have been unable to locate a solution so
far :-(
I want to set-up asterisk so that it mimics a typical answerphone ie.
someone calls in - you call screen in case it's:-
a) a telemarketer
b) someone else you don't won't to talk to
c) someone you don't want to talk to because you are busy
So you listen as they record their message, and if it's
2010 Nov 06
0
Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?
The subject says it all. I'm betting there's a way to do it, but so far
I haven't found the dialplan runestone via web searching.
Thanks.
b.
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
2004 Oct 05
0
sipura 3000 , music on hold (playtones)
hi,
I have some problem with musiconhold or playtones (background,...)
in this context someone dial out thru sipura 3000:
Executing Dial("Zap/1-1", "SIP/sipura3000/054419949|20|m") in new stack
-- Called sipura3000/054419949
-- Started music on hold, class 'default', on Zap/1-1
-- SIP/sipura3000-61fe is ringing
-- SIP/sipura3000-61fe answered Zap/1-1
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
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
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
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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2015 May 11
0
No application 'Playtones'
symack wrote:
> Hello Everyone,
>
> We have most of the modules commented out. Can someone please let me
> know which modules needed to be included for Playtones?
The PlayTones application is in the app_playtones module.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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
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",