garbled.. and
therefore call disconnects when google tries to parse it. Can someone please
let me know how to handle this ?
Also is it a legitimate way of using Google speech recognition and TTS in this
way ? i.e no license issues would arise?
thanks
Sriram
Hi
I want to play streaming music from an internet lp like
http://114.23.245.234:9000 . there will be maximum of 75 callers at a time.
Is combination of SetMusicon Hold and WaitMusiconHold command the best
option ? All callers would be calling on PRI lines. The streaming source
would be from
of mpls
can anyone please suggest
Rgds
Sriram
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
not expert I understand networking. So request you to please guide me
Thanks
Sriram
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
is on a call
??
Requesting for a help
Thanks
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
the VM_NAME for a particular extension
Thanks
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
= 10
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
call-limit = 20
member = SIP/100
member = SIP/101
member = SIP/102
Please help , I m in a total mess .Thanks Sriram
=no
setinterfacevar=yes
setqueueentryvar=yes
timeout = 10
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
call-limit = 20
member = SIP/100
member = SIP/101
member = SIP/102
Please help , I m in a total mess .Thanks Sriram
be IN USE ..also the transfer
event is not getting logged on the queue log file ...am i doing anything
wrong ?
Please help its urgent
Thanks - Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October
with extension
number pointing to the agent who did the transfer
My setup : Trixbox 2.6 with Asterisk 1.4.22
Please help
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
and then pressing ## and caller b agent
ID# still nothing happens ..Can anyone help ?
Thanks in advance
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now
Hi List
Might be a very silly question, I want to make some changes in CDR.C of
Asterisk ( i m using trixbox) . I noticed that cdr.c is present inside the main
folder of svn asterisk 1.4 branch. If i make any changes in cdr.c how do i
update hte changes as i dont see a loadable module cdr.so ?
a working TE420F on another
Dell POweredge T100 machine at a different locaiton ... The DELL tech support
says his PCI express slots are fine ..has anyone encountered this problem
before ? I've tried with 3 different TE420 cards to rule out a defective card
Thanks
Sriram
Hi List
I've a CID lookup hooked onto an inbound route (i m using trixbox) ...it runs
well but it returns the value as CIDNAMECIDNUMBER ... if i just want to
display the CIDNAME [leaving the quotes and CIDNUMBER] .. how can i do it ?
do i have to edit some macro in extensions.conf ?
rgds
Hi All
am using trixbox with call queues..I've set setinterfacevars=yes in queues.conf
and following is dial plan :
[test]
exten = s,1,Answer()
exten =
s,2,Set(FILE_NAME=${CALLERID(num)}-${MEMBERINTERFACE}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,3,Monitor(wav,${FILE_NAME},m)
exten =
Hi
1. I want to record all calls that land to an agent via a queue using a
meaningful name - as of now i name the recorded file on the fly using
{CALLERID} variable so that the file gets stored using the caller id iunder
/var/spool/asterisk/monitor , now if i want to store it as CALLERIDEXTEN
the NAGIOS_USER to root and changed the
ownership permissons on the script also to root..I now get the correct status
on the Nagios interface..
thanks for all your help - Sriram
) What PATH does the script have when run by the Nagios process?
) Are there any permissions issues on the directories
and given the same path in
commands.cfg under objects folder of /usr/local/nagios/etc ... can anyone
please help me out ?
Thanks Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to show
them line by line Up or down) using nagios using below script, but i always get
the status as down and red..can anyone let me know how to read an output from
nagios plugin ? nagios etc is configured already and is
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to show
them line by line Up or down) using nagios using below script, but i always get
the status as down and red..can anyone let me know how to read an output from
nagios plugin ? nagios etc is configured already and is
Hi
I am a premium voice service provider giving some services on IVR to a Telco X
. As my premises is some 10 kms away from that telco , i have taken a PRI
connection (30 DID with 1 hunting/pilot number) from telco Y When a customer
of Telco X dials my short code @Rs.6/- per minute his call
My setup : Trixbox 2.6.1 TE410P running well .:
1. I need to store the CallerId of the PSTN caller with his language preference
so that next time he is played the prompt in his language that he chose the
first time.What would be better - storing his number in the Asterisk DB and
using Dbput
effectively
Thanks
Sriram ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi
I've a requirement for one of my operators for an autodialler for which i plan
to deploy asterisk (I already have 3 asterisk servers on PRI running very well
! ). The scene is like : Asterisk will call a customer and play a prompt that
prompts him to press 1 if he wishes to talk to an
Hi Everybody
I've a requirement for one of my operators for an autodialler for which i plan
to deploy asterisk (I already have 3 asterisk servers on PRI running very well
! ). The scene is like : Asterisk will call a customer and play a prompt that
prompts him to press 1 if he wishes to talk
pop-ups for my agents...i know SIP
extensions can simplify my setup but echo problems scare me also i dont want to
throw my legacy pbx on which i invested heavily..
Can anyone throw some pointers ?
Thanks in advance
Sriram ___
-- Bandwidth
Hi
I am using Trixbox 2.4 and PRI lines..on the CDR i see many calls that have
duration of 0 seconds, but they are still shown as ANSWERED . how come its
possible when duration is 0.00 ? Are the callers billed for such calls ?
Rgds
Sriram
trunk is E1 PRI while i used softphones internally - i dont want my agents to
see the caller id - is their any way to block caller ids from appearing on
softphones ?
Thanks
Sriram ___
-- Bandwidth and Colocation Provided by http://www.api
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via span2)- legacy
pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR option and
they are connected to the agents via the legacy pbx (which is in sync with
asterisk on
hi Robert
followed your points - but problem persists...everything goes well for sometime
but after that - asterisk is unable to dial the pbx...
any more thoughts
thanks
Sriram___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi
-Executing [EMAIL PROTECTED]:1] Dial(Zap/13-1,ZAP/g2/3901) in new stack
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup(Zap/13-1,) in new stack
== Spawn extension (custom-app,1,2) exited non-zero on 'Zap/13-1'
-- Hungup 'Zap/13-1'
this is the
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via span2)- legacy
analog extensions.
my dial plan is like callers dial into asterisk(span1) and they are connected
to the agents via the legacy pbx (which is in sync with asterisk on
span2)the prob is when
) and the pri card of the legacy pbx. but
when i try to make a call to asterisk so that it can send the call to the
legacy pbx using Dial command - it exits saying - CHANUNAVAIL , but if i try to
dial an external PSTN number the call gets thru..
Any help apprecriated.
Thnks
Sriram
;)
Thanks
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
that there is a cdr.so
module that gets loaded - can it help me in anyway
Thanks in advance
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk
on a single call possible in Asterisk ? If yes
how and what additional parameters do i need to get from him
Please assist
Thanks
Sriram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
the asterisk box
should not send the reversal to the billing switch.. only after pressing 1
should the charging begin...I hope am clear now
Any ways to implement this ?
Rgds
Sriram
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, September 09
? or some
additional RAM and processor ?
I;ve been working all along on Dialogic but want to shift to Asterisk as it has
lot of features and just fits in my needs (PBX + IVR in 1 box! ).
Please advice
Thanks in advance
Sriram
___
-- Bandwidth
machine ? or some
additional RAM and processor ?
I;ve been working all along on Dialogic but want to shift to Asterisk as it has
lot of features and just fits in my needs (PBX + IVR in 1 box! ).
Please advice
Thanks in advance
Sriram
___
-- Bandwidth
Hi
I am a premium voice service provider giving some services on IVR to a Telco X
. As my premises is some 10 kms away from that telco , i have taken a PRI
connection (30 DID with 1 hunting/pilot number) from telco Y When a customer
of Telco X dials my short code @Rs.6/- per minute his call
42 matches
Mail list logo
