Displaying 20 results from an estimated 20000 matches similar to: "debug message: Internal timing is disabled"
2008 Nov 26
1
bridging - Didn't get a frame from channel
Hi,
I am having a difficulty with
getting two realtime user?s to bridge on answer. I have managed successfully to
bridge the same two users/channels via the Bridge Manager api command and
confirm that the two communicate directly bypassing the asterisk server (I
confirmed this with Wireshark).
Does anyone have some ideas? I have
put some log entries below.
I haven?t attached my
2010 Jun 13
2
bug with Moh on MeetMe ?
Hello,
The MeetMe application refuses MusicOnHold personalized and skip always in
the default!
Have you any idea how to fix this?
-- Executing [028883899 at default:1] Set("SIP/109.10.214.1-00000002",
"CHANNEL(language)=fr") in new stack
-- Executing [028883899 at default:2] Answer("SIP/109.10.214.1-00000002",
"") in new stack
-- Executing
2006 Jan 14
3
1.2.1 "Silence suppression is disabled" what the hell?
I upgraded from 1.0.9 to 1.2.1.
In 1.0.9 everything worked perfect.
Now, I call in my IVR, and after navigating in menus when I get dialtone
for dialing extension, Sound is choppy and I get bunch of messagess:
-- Silence suppression is disabled (option_silence_suppression=0
chan->timingfd=30)
-- Silence suppression is disabled (option_silence_suppression=0
chan->timingfd=30)
-- Silence
2007 Mar 19
0
1.4.1 - T38 Pass Through - Seeing some odd errors but the fax works.....
Hello List -
Here's the setup:
Mediatrix 1102 ATA (t38enabled) <--> Asterisk 1.4.1 <--> IP <--> SIP GW <-->
TDM
The T38 call comes up perfect - I see the initial invite, followed by G711,
Re-Invite, T38 establishes, Fax Completes, T38 Stops, Call Down.
here's the problem - I see the following in my console:
[Mar 19 05:09:38] WARNING[4745] chan_sip.c: Can't
2009 Oct 02
0
Sending a DTMF remotely with PlayDTMF problem.
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2009 Oct 03
0
Problem sending a DTMF remotely. Please need help...
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2009 Oct 06
0
Problem sending a DTMF remotely. Please need help!!!
Hello, how are you?
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended
2009 Oct 05
1
Problem sending a DTMF remotely. Please need help!!
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2009 May 26
0
No Voice - only "noisy audio"
Hi Folks,
I'm trying to use my mobile as a trunk via bluetooth - calls done in a
softphone go thru GSM network and calls destinated to my mobile are answered
at the softphone.
I have asterisk configured to do so but I'm facing an issue - Audio is
audible but it?s not intelligible. I feel like the audio is breaking.
Below is the asterisk log. I also get lots of ?hci_scodata_packet: hci0
2007 May 09
1
Replaces header
I'm tying to use park and announce for call park on Asterisk 1.4.2 but
I'm having trouble getting it to work properly. This use to work with an
older version of Asterisk.
A telephone on the PSTN calls an IP phone. The IP phone is assigned
extension 3-8396. 3-8396 answers the call and attempts to perform a
blind transfer to x700, the parking lot number. The transfer gets to
Asterisk,
2010 Jun 12
1
MeetMe problem
Hi Guys,
sometimes if one caller or many callers are in a meetme Room and a new one join the room,
then he or another caller into the same room where kickt from the room.
It's very strange for me and in logs (full) I can't see anything. is it possible to log more from meetme.c ?
can anyone help me and maybe someone has also the problem as i and have an solution.
I use:
asterisk-1.6.2.7
2010 Oct 11
2
user number in conference
Hey,
i forgot to ask, how can i get the user number from a caller he is in a conference, i don't find a variable to us this for the current channel.
Only the command "meetme list <roomnr>" shows the usernumber, but i can't use this output.
Thanks.
Daniel
2006 Jan 15
3
MoH trouble with latest bristuff (0.3.0-PRE-1f)
Hi,
I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When
I activate music-on-hold on a SIP-to-SIP connection, the music sounds
like in a fast-forward play mode. On the *-console I can see much lines
like this:
-- Silence suppression is disabled (option_silence_suppression=0
chan->timingfd=18)
What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but
2008 Dec 10
0
Replace music-on-hold on MeetMe with ringing sound
Date: Mon, 23 Jun 2008 08:00:08 -0400
From: "David Backeberg" <dbackeberg at gmail.com>
Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with
ringing sound
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<3de056a30806230500k7e66185l7bfe473ed398ebf6 at
2006 Jan 30
0
Meetmee weirdness
I have several instances where conference calls are not being torn down
appropriately. My CDR shows 3000 minute calls, which are coming in on PRI.
I know that the calls aren't really lasting that long. What could be
causing this? IN fact, here is what shows now:
asterisk*CLI> meetme
Conf Num Parties Marked Activity Creation
138 0000 N/A
2005 Jan 12
2
T1 Timing Slips
Does anyone know how to monitor * to see if they are receiving timing slips
on a span connected to a T100P card? I am seeing b-channel restarts quite
often and also getting "No D-channels available" warnings from time to time.
Yesterday I had all the b-channels crash during a MeetMe Conference. Not
good! This PRI is connected to an Avaya Definity PBX that is onsite and
located in the
2009 Jul 13
1
Trouble with originating a call through Asterisk Manager Interface
I am doing a little application to originate a call through Asterisk via AMI
(Perl Asterisk::Manager).
It logs in successfully, does an originate command with
Exten: 0020 (which is set up to answer and wait for 60 then hang up)
Channel: SIP/5101234567 at test-host (which comes to my desktop machine also
running Asterisk).
At the target machine I see only a CANCEL to which it immediately responds
2004 Jul 06
2
ztdummy running, but moh & meetme don't work
Any thoughts on the following?
I am running asterisk from CVS (downloaded yesterday's
version, just to be sure) on a test system with no
digium cards in it, so I have installed ztdummy (see
logs and screenshots below) as a timing source.
When I call the music on hold extension from a Sipura
Sip connected analog phone, I hear nothing and start
getting
Warning[98310]: chan_sip.c:674
2005 Jan 19
0
MeetMe MusicOnHold Volume
I've got a simple MeetMe conference configured using Asterisk 1.0.3 on
Gentoo. I'm using zaprtc for timing from the bri-stuff package.
extensions.conf
exten => 37455,1,NoOp(Drill Squad Conference)
exten => 37455,2,Monitor(wav,drillsquad-37455,mb)
exten => 37455,3,MeetMe(37455,pMs)
Now, when I enter the conference as the first call, the MusicOnHold
plays, but it's blasting
2014 May 09
3
authoritative sql definitions for Asterisk Realtime Architecture ARA
I am trying to find where the authoritative sql definitions for Asterisk
Realtime Architecture ARA are located. I have found many locations but each
and everyone seems to be different.
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html
http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example
Files included with the distribution: