Displaying 20 results from an estimated 1000 matches similar to: "Cant Play gsm file"
2007 Jan 16
2
command like break ore exit in the dialpan
Hi
i have a similar dialplan:
exten => 99,1,Gotoif(....?2:3)
exten => 99,2,Meetme(100)
exten => 99,3,Meetme(100|options)
i'd like to do something like:
exten => 99,1,Gotoif(....?2:4)
exten => 99,2,Meetme(100)
exten => 99,4, ... exit ...
exten => 99,3,Meetme(100|options)
How can i exit the dialplan?
I won't use meetme twice!
Thanks nik
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
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 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
2007 Nov 12
3
No sound from playback and voicemail
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played, but I hear
nothing at all.
Here is as simple example:
[monkeys]
??? exten => 99,1,ANSWER()
??? exten => 99,2,PLAYBACK(tt-monkeys)
??? exten => 99,3,HANGUP()
The phone
2004 May 14
7
What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten => 99,1,VoicemailMain(${EXTEN}@inside)
exten => 99,2,Hangup
Paul Mahler
pmahler@signate.com <mailto:pmahler@signate.com>
<http://www.signate.com/>
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
2003 Jun 19
1
Unable to find a path
Hi!
I just installed Asterisk 0.4.0 with all the default options, and the
configuration samples it has. When I try to dial from an h323 client
(gnomemeeting) I get this message on the messages file:
Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream):
File demo-congrats does not exist in any format
Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):
2003 Jun 13
1
strace shows that files are not accessed
strace on file access in asterisk shows that * is not even attempting to
access the voice files.
If I *manually* load app_playback.so, app_macro.so, and then
pbx_config.so, I they will load and I get a dialplan. Ok, that's a
problem -- autoconf is clearly not working, or there's some other
related issue.
So I try to use the demo and do "dial 500". This should connect and
2004 Nov 20
1
Asterisk dead but pid file exists - gdb asterisk core.13089
Dear ALL,
Any clues or tips for the following gdb messages.
[root@localhost asterisk]# uname -a
Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct
29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux
localhost*CLI> show version
Asterisk CVS-HEAD-09/22/04-11:19:09 built by
root@localhost on a i686 running Linux
[root@localhost asterisk]# gdb asterisk core.13089
GNU gdb Red Hat Linux
2014 Dec 12
1
Corrupt MixMonitor recordings - .gsm format
Hi all
Asterisk 1.8.11.0 on Centos 6.5
My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom,
South Africa). Unlicensed G729 codec version on server.
75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes
into the recording.
The server has been up for 7 months beforehand with no problems with
recordings to .gsm format files.
I noted
2004 Aug 24
2
Voicemail & "Couldn't read username" error
Hi,
I have Asterisk running with the VoiceMail. Using the latest CVS. I have
my extensions.conf setup so that if a SIP caller dials *99 the
VoicemailMain() as follows:
exten => *99,1,Wait(1)
exten => *99,2,VoicemailMain()
A couple days ago I installed the MySQL/Voicemail support described at
http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some
reason
2003 Apr 04
2
chan_h323 problems....
I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc.
I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs
pwlib
openh323
gnugk for h.323 gatekeeper
2004 Jan 18
2
Nufone not taking GSM CALLS
Is nufone having problems taking gsm calls today
i had some issues dialing overseas to call my folks.
here's snip of what the console displayed
-- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack
Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1
2003 Aug 07
2
Problem -ATA-711-723-Oh323-Asterisk
Hi List,
I am facing the reverse problem as stated here.I am
using ATA 186 to make
and recieve call to * through OH323 driver.
When I use G711 codec in the ATA to make call then
then as soon as i dial an
extension the * crashes with 'segmentation fault'.
But the same scenerio works fine when i use 723 codec
in the ATA .I can dial
the number and extension very well/(I have 723 support
in
2007 Nov 12
0
No sound from playback and voicemail (Atis Lezdins)
Hello,
>> > I can talk to other SIP phones and via ISDN to the outside, but I
>> >don't hear playbacks or the voicemail messages.
>> > Asterisk show in the cli, that the corresponding files are played,
>> >but I hear nothing at all.
>> >
>> > Here is as simple example:
>> >
>> > [monkeys]
>> > exten =>
2003 Sep 18
2
Adpcm quality
Please, try
exten => 99,1,Wait,1
exten => 99,2,Record,/tmp/pcmfile:pcm
exten => 99,3,Wait,1
exten => 99,4,Playback,/tmp/pcmfile
exten => 99,5,Wait,1
exten => 99,6,Record,/tmp/voxfile:vox
exten => 99,7,Wait,1
exten => 99,8,Playback,/tmp/voxfile
(put your own extension).
Pcm recording is OK, playback is OK.
Adpcm recording is noticeably worse. Adpcm playback is very
2004 Nov 26
2
Execute a script upon registration
Is it possible to execute a script upon successful registration and
authentication of a SIP device in Asterisk? For instance, have a script log
all successful registrations in a database or authenticate users instead of
using the secret=password in the sip.conf file? Thanks -
--
Brian Wilkins
Software Engineer
brian@hcc.net
Heritage Communications Corporation
Melbourne, FL USA
2007 Jun 29
1
Fwd: Call Wainting dysfunctions
I am trying to implement a Centralized Call Waiting System. I have red
some document about asterisk group features to manage group and
category of a sip channel.
I have done a lot of test about it but always it doesn't work
correctly if I transfer the call.
This is the macro code I use for inbound calls.
[macro-test]
; ${ARG1} - technology something like SIP
; ${ARG2} - resource.
2007 Jun 19
2
make config
Hello everybody, when I run "make config" I have this error:
install: cannot stat `init.asterisk': No such file or directory
make: *** [config] Error 1
I don't understand.
For what is "make config"? to put on /etc/init.d/?
Thanks for all
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2003 Jun 24
3
Compiling Asterisk under Yellow Dog
Hi,
I am trying to compile Asterisk under Yellow Dog 3.0 distributionn.
I am getting an error
gcc -shared -Xlinker -x -o codec_gsm.so codec_gsm.o -lgsm
/usr/bin/ld: cannot find -lgsm
May be I need packages that my distribution does not include?
What do I need to download to get it compiled?
Thanks
Serge
_________________________________________________________________
The new MSN 8: smart spam