Displaying 20 results from an estimated 700 matches similar to: "Sounds"
2008 Oct 13
1
Need help for debuging
I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash?
It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened.
This asterisk is using as ACD for over hundred agents.
#> thread apply all bt
........
........
Thread 6 (process 20135):
#0
2007 Mar 02
1
cmd page crashes Asterisk SVN-branch-1.4-r57207
Group
I'm having some trouble with asterisk and the page cmd.
Any help would be great!
This is what's in my extensions.conf
exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten => _**2,2,Page(SIP/36651)|d
exten => _**2,3,Hangup
CLI output
********************
Connected to Asterisk SVN-branch-1.4-r57207 currently running on
VoIP-PBX (pid =
2006 Nov 02
0
sound-files not playing?
Hi all!
In my extensions I have the following:
exten => 999,1,Answer()
exten => 999,2,PlayBack(beeperr)
In /var/lib/asterisk/sounds/ I have both beeperr.gsm & beeperr.ulaw,
both with '-rw-r--r--' permissions.
when I dial extension 999 I get:
************************************
-- Executing Answer("SIP/asterisk.domain.com-081477a0", "") in new stack
2005 May 30
2
R: R: AT-320 + supervised transfer
The procedure that will do asterisk is very nice ;) but whe it was available ?
Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why.
Here my sip.conf for the phone, can u say me if there is somethingh wrong ?
[2391]
type=friend
username=2391
secret=2391
language=it
host=dynamic
context=intern
dtmfmode=rfc2833
2011 Oct 19
1
Asterisk call transfers not working
Hello:
We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0
running. Everything seems to be ok but call transfers. This is the issue:
*A, B, C and D are in FXS ports*.
1) A calls B. B anwers.
2) B tries to transfer the call to C dialing *2 (code for attended
transfer).
3) A hears MOH. B dials number C.
4) Asterisk says the dialed number is incorrect or non existing.
We tried
2005 Jun 29
0
(no subject)
Hi, I installed mpg123 v0.59r without error and defined as defaut folder
/var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem
ok, but i cannot hear music. I'm using asterisk 1.0.8
*CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new
stack
-- Called 2391
-- SIP/2391-79a0 is ringing
-- Saved useragent "PA168S" for
2005 Jun 29
4
Music oh hold
Sorry, i also tried this:
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold(default)
and i got this result:
*CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack
-- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack
Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class
2005 Jun 30
3
R: Music oh hold
This is my musiconhold.conf and my folder:
root@voip:/etc/asterisk# less musiconhold.conf
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => mp3:/var/lib/asterisk/mohmp3,-z
;unbuffered => mp3nb:/var/lib/asterisk/mohmp3
;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3
; Note that the custom mode cannot handle escaped parameters
2008 Dec 05
2
IAX trunk mixing
hi,
i have a problem, and i am completely stuck with it, i hope someone can
point out where is my config wrong.
I have three server, connect together with IAX trunking. The server are
at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia
(10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the romanian
server, i dial a hungarian telephone number, the call goes to the
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 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
2006 Apr 11
0
log messages...
Hi,
Gere are some messages that sometimes show up in my Asterisk logs... If you help me out to solve them, I could make a list of all know warning messages so that we can publish in the wiki or somewhere else!
- "res_features.c: Did not read data." - on Google, the only reference to this was in Russian :(
- "Asked to transmit frame type 64, while native formats is 256 (read/write
2007 Jul 23
0
app_changrab, replacement for meetme and conference: returning to dialplan
Hi all,
there is an application called changrab with quite interesting capabilities:
http://www.freeswitch.org/asterisk_stuff/app_changrab.c
I think it is available in the 1.4 version by default!?
This application can connect to channels which are already UP. The only
possibility AFAIK to connect channels after they are UP are the well
known conferencing applications meetme and conference. If
2010 Sep 29
0
Successive Dial apps give hang up within 30s!!
Hi All,
I am using an Asterisk 1.6.2.6, and when I use this part of the dialplan:
exten => 8355,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN},18,tTWwr)
exten => 8355,n,Dial(IAX2/8366,48,tTWwr)
(i made that simple to exhibit issue)
I got just 1 ring in 8366 extension before it hangup, what i noticed is the
total time spent on ringing is 30s that means if i use 12s in the first dial
i get 18s left
2004 Aug 26
0
Out Dial Problem
Dear All,
I just setup the Asterisk with E100P which it's no problem in Dial In but I
have problem when outdial. The connection method is like this :
E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP
\-----> Analog PHone
Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect,
Trying,
2007 Dec 04
0
Queue App - crash (1.4.15)
This is the core trace
(gdb) bt
#0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6
#1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828
"default", interpclass=0x0)
at res_musiconhold.c:646
#2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of
bounds>,
interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614
#3
2016 Jun 30
3
how to join 2 channels using AGI/AMI
sorry for top-posting, the two topics started with 2 different reason
subject, but then we finished on the same problem.
btw,the 2 show channel are reported above:
the channel with DTMF working
kcenter*CLI> core show channel SIP/pbx2-000004b9
-- General --
Name: SIP/pbx2-000004b9
Type: SIP
UniqueID: 1467323106.1275
Caller ID: xxxx
Caller ID Name: xxxx