Displaying 20 results from an estimated 100 matches similar to: "issue with speech in IVR"
2013 Jul 04
4
Digium Analog card and Asterisk
Hi
I just bought some digium analog cards and I would like to build an IVR
system for my customers.
However I am googling and googling , I didn't find any blog and instruction
for beginners like me. So I come here for help. Any tips or blogs will
help.
Regards,
Hua Jie
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2009 Jul 23
5
Music on hold based on user
Hi
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play music
1, if another from Technical Support put on hold the call play music 3,
something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
Thanks
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2013 Jul 25
1
asterisk and IVR
Hello list,
i need your help about the IVR please
i have asterisk 1.4 installed and i configure an IVR like below
exten => 529,1,Ringing()
exten => 529,n,Wait(4)
exten => 529,n,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}welcome)
exten => s,n,WaitExten(5)
exten => s,n,goto(home,s,1)
exten
2006 Mar 30
3
Is mount_smbfs broken in 6.1-PRERELEASE?
Anyone know if mount_smbfs is broken in 6.1, I'm trying to run this:
"mount_smbfs -I 192.168.1.2 //nbritton@192.168.1.2/music2 /mnt/network/music/"
And then it asks for my password, I type it in, and then I get this error:
"mount_smbfs: unable to open connection: syserr = Authentication error"
I've had this same problem on another 6.1 box too... I can run this
same
2004 Aug 06
1
Why doesn't yp.icecast.org show my stream?
Jack Moffitt <jack@xiph.org> writes:
> What are you trying to do that you get that error?
I sent an improperly formatted URL to your server. I've corrected this
since. =)
> id=69 is the normal success code. You'd get -1 if it failed I think.
> What's your ip address? I'll go check the logs.
The ip address of my streaming server is 216.133.255.2
> Other
2016 May 12
0
3WK needs help with ogg comments uppper and lower
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"
"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en"><head>
<title></title>
<meta http-equiv="content-type" content="text/html;charset=utf-8"/>
2007 Jan 30
3
musiconhold restarts for every extension
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
;music starts
exten => 902,1,Dial(SIP/phone1@proxy.com|5|m(mymusic))
;music starts again
exten =>
2007 Mar 13
0
MusicOnHold stops after upgrade from 1.4.0 to 1.4.1
Hello
I have following problem.
After upgrade from 1.4.0 to 1.4.1 my musiconhold stops immediately after
start.
Bellow some logs from 1.4.0 and 1.4.1 (same configs and situations)
First, the one from 1.4.0 (everything works)
[Mar 12 13:44:00] -- Executing [s@info800:1]
SetMusicOnHold("SIP/1036690-b74004b8", "mymusic") in new stack
[Mar 12 13:44:00] -- Executing
2015 Mar 12
5
chanspy for group extension
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a ?crit :
> hello list,
>
> i use the code below
>
> [macro-chanspy]
> exten => s,1,Authenticate(${ARG1})
> exten => s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
2013 Mar 25
7
question about zapata.conf
hello list,
i have a question related to zapata.conf,if i do any change in zapata.conf
i must restart asterisk or just i restart zapata ,and how to do .
?service zaptel restart? or there is any other command
Thanks and regards
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2010 Jun 18
6
asterisk issue
Hello,
I have a problem in Asterisk 1.4 each day I need to restart *asterisk
service asterisk* restart in order to unblock the calls
My question how can I do in order to check the issue, and if there is any
tool or log?
Thanks and regards.
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2015 Mar 26
2
call between snom 300 and aastra 6731i
hello list
i need your help please regarding an issue with snom300 and aastra6731i
using asterisk
11.13.0 asterisk
snom 300 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and aastra ====ok inbound and outbound
the calls between x-lite and snom300====> ok inbound and
2015 Mar 12
2
chanspy for group extension
thank you so much it work
you must add 1 like below
[app-chanspy]
exten => _0071XX,*1,*Macro(chanspy,1234)
exten => _0072XX,*1,*Macro(chanspy,5678)
exten => _0073XX,*1,*Macro(chanspy,8910)
best regards.
2015-03-11 19:48 GMT+00:00 Carlos Chavez <cursor at telecomabmex.com>:
> On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
>
>> hello list,
>>
>> i use
2011 Feb 14
3
issue with some numbers
Hello all
I have a small issue with some mobiles numbers when I call these numbers
using asterisk I have all the time answer machine. But when I call these
numbers using my mobile or another phone there is no problem.
Any help will be appreciated
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2011 May 30
3
please help
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten => _0678922645.,1,Set(CALLERID(number)=520460587)
exten => _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten => _067892264*5*,2,Hangup()
i can not call my
2011 May 07
3
record call from iax to sip
Hello List,
i need to be able to record the call transferred from iax extension to sip
extension
when i call the sip extension from the IAX extension i can record the call
without any issue
but when i receive a call from customer in IAX and i transfer this call to
SIP client
the conversation between customer and IAX client is recorded but the
conversation between customer and sip extension is
2011 Jan 31
2
save the calls with asterisk
Hello All,
I have asterisk installed in our call center and i want to know how to do in
order to save all the calls (inbound and outbound) if there is any tool
Thanks in advance
Kind Regards.
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2013 Mar 26
2
WARNING[28151] from CLI
Hello,
i have all the time this warning i use asterisk 1.4 all works without
issue i don't have any problem (i can use the inbound and outbound calls
without issue)
i just want to know what is this WARNING
thanks and regards
WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available!
Using Primary channel 140 as D-channel anyway!
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An
2011 May 19
2
click to call with php
Hello,
i have asterisk 1.4 installed and i want to use click to call in order to do
an outbound call
if there is any php code in order to do this operation
thanks and regards
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2015 Mar 11
2
chanspy for group extension
hello list,
i use chanspy with the code below
[app-chanspy]
exten => _007.,1,Macro(user-callerid,)
exten => _007.,n,Answer
exten => _007.,n,Authenticate(1111)
exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten => _007.,n,Hangup
i have a question related to chanspy
i have created extension from 100 to 300 and i will give the permission
with group of extension
i want to use