Displaying 20 results from an estimated 20000 matches similar to: "SIP transfer doesn't stop music on hold"
2003 Apr 15
2
RE: [Asterisk-Dev] Several patches, including recording and music -on-hold
Mahmut, excellent summary :-). I look forward to your next update.
One little thing,
In the manager events that show start/stop monitoring, can you please include a field
that indicates the filename(s) to which the monitoring was written?
Thanks,
Ben
-----Original Message-----
From: Fettahlioglu, Mahmut [mailto:Mahmut.Fettahlioglu@oa.com.au]
Sent: Tuesday, April 15, 2003 5:17 AM
To:
2007 Jul 11
2
Music on hold stops on blind transfer
Asterisk 1.4.6 at FreeBSD6.2-RELEASE
Client hears pure silence when waiting for call answer. Music on hold stops
when transferer pics a number and client doesn't even hear ringing.
Is this normal behaviour? How to change this?
Log says everything, MOH should stop after call pickup, not before Dial.
-- Executing [113 at firma:1] Dial("SIP/zytek-08737000",
2003 Jul 30
2
Call Transfer, Budgettone 100
hi,
can someone who has used Budgettone phones tell me how to do the
following:
an incoming call comes in and is answered by the receptionist.
she need to put the call on hold, speak to whoever the call is for,
and either (after that) pass on the call, otherwise speak again to
whoever was on the call and hang up ..
so far i've got as far as a blind transfer by pressing transfer button
and
2003 Apr 03
0
Music on Hold for SIP
I posted a message a little while ago but got no response (that I can
recall), I've also seen other people mention this issue.
Basically, when you have music on hold, it doesn't play the music on hold,
the debug info shows it is starting and then stops straight away..
# My extensions.conf ...
exten => s,1,Answer
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,10
exten
2004 Sep 15
1
Transfer / Music-On-Hold
Hi All,
I have what IMHO is an interesting issue. I'm using Cisco 7940's with
the 7.2 SIP load and Asterisk CVS-HEAD-09/10/04-10:11:46. Everything is
working great so far, except one small issue.
When a user presses the 'Trnsfer' soft-key, dials the other extension,
and presses 'Trnsfer' again, before the other party picks up, hold music
for the original caller
2010 May 26
1
Getting "ghost" transfer or music on hold
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Tahoma">Hi Everybody,<br>
<br>
I´m getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ...
In some calls, i get an atxfer or musiconhold in the middle of
2006 Apr 18
2
HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again
Hi all,
I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a
call and i press Hold button, the other party starts listening Music on Hold
but then when i press the button again to get the call back it doesn't work!
I've checked asterisk CLI:
-- Stopped music on hold on Zap/1-1
-- Started music on hold, class 'default', on Zap/1-1
--
2004 Jun 29
0
Play Music on hold until a ZAP channel frees up.
[answeringsvc]
exten => 0,1,Wait,1
exten => 0,2,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},r)
exten => 0,3,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},mr)
exten => 0,103,Goto(0,3)
exten => 0,104,Goto(0,3)
This should call 713-555-1212. If there are no ZAP lines available it
should kick back around and play music on hold until a zap line is
available, correct? I'd like the
2007 Oct 22
0
bristuff: music on hold but no dialoptions tT defined.
Hi,
Iam dialing from NT ptp to SIP provider.
Sometimes Asterisk is doing music on hold but there are no options like t or T
in the dial command. As an result the channel got lost and an Hangup occurs.
Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card.
Any solution for this?
Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered
Zap/8-1
Oct 22 11:20:23 VERBOSE[29983]
2003 Sep 24
10
SIP / GrandStream Configuration
Hi there!
I installed the BudgetTone (GrandStream) on my LAN without any problems.
Then, I moved it to another location using a D-Link NAT.
I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address
of the BudgetTone.
When I receive a call on my Asterisk, it would ring my FXS as before.
However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in
the log).
The
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
2006 Jan 07
1
choppy music on hold - only on PRI PSTN
Hello to all.
I do not know what is causing choppy music on hold when call comes in
through E1 card (PRI).. but this channel info is somehow strange.. We use
Alaw over PRI (and I think it's format number 8),
But why is WriteFormat at 2 ?????
Thanks!
show channel Zap/1-1
-- General --
Name: Zap/1-1
Type: Zap
UniqueID: 1136667936.0
Caller
2003 Apr 17
1
timeout music on hold or ring tone
Is any way to limit music on hold (or ringtones) to specified time ? I
need it to play it ~ for 7 seconds .
How to do this ?
in dial plan i have:
exten => _021XXXXXX,4,Dial,Zap/1/BYEXTENSION||r
when go to this extension it rings once!
and then asterisk say :
-- Zap/1-1 answered Modem[i4l]/ttyI0
and it stop ringing ;) becouse mean that other end is ringing :) ..
BUT when the other
2004 May 04
2
Dial zap and music on hold
i tried using music on hold option in the dial command
exten => 7777,1,Dial(zap/1/7777,20,m)
when someone calls me and i picked up the phone, the call will
be suddenly dropped. however, if i use a sip client instead of
zap (also changing the dial statement to sip), i can answer the
incoming call without a problem.
is this a known bug?
(asterisk cvs 05-03-04 using RedHat v9 on Via mini-ITX)
2006 Jun 07
1
Music On Hold not working with new 1.2.7.1 install
I have followed the instructions provided at:
http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf
including installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.
However, no sound is heard and I get this message from the CLI when accessing MOH:
-- Started music on hold, class 'default', on channel
2003 Aug 28
6
SIP and ECHO
Hello,
I have read the information on echo and SIP in the FAQ and I have
scoured the mailing list for possible solutions, but as yet I have not
been able to get rid of this echo.
I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed
into an asterisk server. If I call between the Sip Phone
(Budgettone-100) and the 4 FXS ports everything sounds great. If I call
out to the PSTN
2003 Apr 14
1
RE: [Asterisk-Dev] Several patches, includin g recording and music -on-hold
Hi Wade,
Sorry for replying so late. I had been sucked into other tasks for a while
and only now can catch up with the list.
> When I dial my iaxtel number from my extension on channel
> Zap/15, I get two
> files recorded in /var/spool/asterisk/monitor:
>
> Zap-15-1-in.wav and Zap-15-1-out.wav and they sound fine.
>
> When I dial again, it overwrites the same two files.
2003 Apr 14
0
RE: [Asterisk-Dev] Several patches, including recording and music -on-hold
Mahmut,
First of all, I'd like to reiterate what a great patch this has been.
I'd also like to voice support for having an option to mix the files on
the fly, and name them uniquely. While I was able to smoothly put the
files together with soxmix, I see on the fly mixing as hugely beneficial
to an automated solution to saving/delivering the messages without
intervention.
One feature,
2003 Apr 15
0
RE: [Asterisk-Dev] Several patches, includin g recording and music -on-hold
Thanks Ben, Adam and Petr for the feedback!
So currently things that need to be done for the Monitor resource are:
1. Name files uniquely. Adam, your naming suggestion is great. I think we
should stick with that, with a minor change: I don't think we should put
destination channel name in the file names. In some instances there will be
no destination channels (plain IVR: play, record, dtmf),
2004 Jun 30
1
Can't transfer with Zap and SPA-2000
I am having trouble getting transfers to work when a zap channel is
part of the call. I have a couple SPA-2000's and some X100P cards as my
setup. This is what I'm trying:
Dial number from phone:
-- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack
Currently on call:
-- Called 1/#######
Press flash to place call on hold with SPA-2000:
--