Displaying 20 results from an estimated 500 matches similar to: "Corrupt MixMonitor recordings - .gsm format"
2006 Jan 14
0
codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
Hi guys,
Anyone seen something like below(see below the line)?
Machine P2 w/512MB RAM
Debian (testing) ; kernel 2.6.12-1-386
asterisk 1.2.1-n-all incl. astcc
For many months now I went through * 1.07, 1.09 and never
saw something like that. Even with 1.2.0, a month now,
at the beginning everything was fine, and suddenly
"codec_gsm.c:194 gsmtolin_framein: Invalid GSM data" thing
2004 Apr 20
3
IAX clients are Unmonitored / UNREACHABLE
We have a problem with our iaxclients.
Our asterisk runs on a public host with debian and many of our IAX2 clients
are natted.
The iax.conf looks like:
[23456]
accountcode=123
type=friend
context=user
auth=md5
secret=xxxx
username=23456
callerid=Testuser 1 <23456>
notransfer=yes
host=dynamic
The cli command IAX2 show peers shows all clients as unmonitored
CLI> IAX2
2003 Oct 31
1
Problems with SIP
I'm new to Asterisk, but, Managed to get it working for outound calls from
my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems
with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In
fact, voice mail won't even work..
This is a snippet of what I'm getting when I try to call the ATA
-- Executing
2009 May 21
0
1.4.24.1 -> 1.6.0.9: segfault
I'm testing an upgrade of an i686 production machine running 1.4.24.1 to
1.6.0.9. I've installed dahdi-linux-2.1.0.4.
But:
asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
2004 May 21
0
unable to use EXEC in AGI
dear list
if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain
-- AGI Script Executing Application: (VoiceMailMain) Options: ((null))
May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error
reading:
Resource temporarily unavailable
May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205
__adsi_transmit_messages: Un
able to send CAS
May 21 04:25:10
2004 Oct 05
0
Just getting started with Asterisk
Hi list,
I have been looking around for a while now, and cant seem to get to the
bottom of my problem.
My setup is that I have a separate SIP server that servers my SIP
subscribers, and I want to use Asterisk purely for voicemail for now.
So I set up a common SIP extension at my SIP server, and made Asterisk
register it, so that normal users can forward calls to that common
extension, and
2005 Jul 03
0
no sound. "Failed to write frame"
Hi all,
Couldn't find a place to search the list archives...
I'm having issues in getting any sound using a fresh asterisk install
and a SJPhone to connect to it. I went by the instructions pointed at
the "10 minute guide", located here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart.
I installed it on a Slackware 10.1, by using no more than "make
2005 Jul 04
0
no sound. "Failed to write frame" (2nd post)
Hi all,
Couldn't find a place to search the list archives...
I'm having issues in getting any sound using a fresh asterisk install
and a SJPhone to connect to it. I went by the instructions pointed at
the "10 minute guide", located here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart.
I installed it on a Slackware 10.1, by using no more than "make
2007 Sep 16
0
Problem with asterisk 1.4.11 and playing files to meetme conference
I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm
using a call file to connect a meetme conference to an AGI script which
plays files using the stream_file method. I have four files which should
play in sequence, though only the first two files actually play. I get
these errors in the CLI:
[Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio
bytes: 276 Buffer
2007 Aug 06
1
Cant Play gsm file
Hi,
i am having problem on playing asterisk sound file on my new installed
asterisk..
i have the following extension , if i call from any SIP / IAX phone
playback or voicemail doesnt play anything .... but when i dial 102, I
hear the MP3 music ..
exten => 99,1,Answer()
exten => 99,2,Playback(prepaid-welcome)
exten => 99,3,Hangup()
exten => 101,1,VoiceMailMain()
exten =>
2004 May 19
1
Old sound in new call.
Hi,
I have a problem that I just can't figure out how to solve.
I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in *
I get the demo-greeting, listen for a few seconds and hang up.
I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should.
Right now I have removed all codecs but codec_gsm.so
2018 Jan 11
0
R-hts
thanks jeff and jeremie,
i am attaching 40 rows of the data, randomly picked from the large table.
the vars are - entity (1-46, with some missing IDs not included due to
missing data), group (1/2), sub group (1/2/3/4), year (2002-2016), y, x1
and x2 - large values included due to size of players - (may not be
considered as outliers as they constitute the sample and are important
countrywide
2004 Jun 02
1
oh323: Failed to create smoother
Hello,
I tried to get the oh323 drivers running. The driver loads, but as soon as a
H323 voice communication should be started, following error occurs:
-- Executing Playback("OH323/R1", "invalid") in new stack
Jun 3 01:26:20 ERROR[294931]: chan_oh323.c:1933 oh323_write: OH323/R1: Failed
to create smoother.
Jun 3 01:26:20 WARNING[294931]: file.c:539
2005 Sep 21
1
Addendum to Problem with Queues question
Here is the full "transaction"
-- outgoing agentcall, to agent '1001', on
'Local/3044@local-4fee,1'
-- Called Agent/1001
-- Executing Macro("Local/3044@local-4fee,2",
"sipline|3044") in new stack
-- Executing Dial("Local/3044@local-4fee,2",
"SIP/3044|20|t") in new stack
-- Called 3044
-- SIP/3044-ea92 is
2007 Jul 05
2
sometimes calls drop during attended transfer
Hi,
I'm testing attended transfer with 3 SIP phones. I noticed about 10% of
my transfers make the call drop and I get this on my log:
Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback:
Failed to write frame
-- Playing 'beep' (language 'it')
Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer:
Failed to play transfer sound!
Moreover, every
2009 Aug 14
1
play prompt after hanup
Hi,
Can I play a prompt after hanging up a call? I have tried below but failed.
...
exten => s,n,Dial(SIP/1234)
...
exten => h,1,Playback(demo-instruct)
-- Executing [h at macro-safedial:2] Playback("SIP/3601-09856bf0",
"demo-instruct") in new stack
[Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback:
Failed to write frame
--
2006 Jan 18
1
Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds
Hi all!
This is my VoIP network scheme
H323EndPoint ----- --- GW H323/SIP-IN
-- -- SIP Phone
| |
(Sipquest) | |
|
| | |
2013 Mar 04
2
Asterisk 11 - How to trim the number of modules to minimum ?
Hi,
I've got a brand new Asterisk 11 setup for which I would like to keep the
number of loaded modules to a minimum.
My goal is to this setup in a pure SIP environment, for switching incoming
calls to outgoing tSIP trunks.
When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an
incoming SIP call with a Playback app.
When I leave autoload=no in /etc/asterisk/modules.conf, it
2013 Feb 21
2
Playback on h exten
Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup.
I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got:
-- Executing [301 at from-test:1] Dial("SIP/300-00000045", "SIP/301,60,rjtTg") in new stack
-- Called SIP/301
-- SIP/301-00000046 is ringing
2004 Dec 06
1
Console as extension problems
I'm trying to set up the console as an extension (so I can set up overhead
paging), but I can't seem to get it to work. When I call my paging extension,
I get an error that it can't open the device:
-- Executing Ringing("Zap/9-1", "") in new stack
-- Executing Dial("Zap/9-1", "Console/dsp0|18|A(new/whistle)") in new stack
<< Call