Hi All
I am having a strange problem when I call from 1 RTC
Client to another without Asterisk in between
everything use to be fine but when asterisk is there
as a Registrar a problem use to occur in many calls,
Caller can hear the voice of the receiving side
but the receiver cant be able to hear
i have tried
rsgain=100
txgain=100
recording volume improved but still not good
--- Steve Totaro [EMAIL PROTECTED]
wrote:
Hi All
I have a call center working on 8 FXO Channels,
everything working fine except one little problem,
I
am using asterisk queues with
monitor-format =
Hi All
I have a call center working on 8 FXO Channels,
everything working fine except one little problem, I
am using asterisk queues with
monitor-format = wav49
and
monitot-join = yes
asterisk is recording all conversations but the
problem is that the volume of Zap Channel is too low
in most
I am using EWSD's PRIs and I am not having this
problem my configs are
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = us
defaultzone=us
Zapata.conf
[channels]
language=en
context=ext-acd
switchtype=euroisdn
signalling=pri_cpe
echocancel=yes
echocancelwhenbridged=yes
theres nothing between Lucent Max TNT and Asterisk box
both are using same IP Class network and both are
connected on same switch
--- [EMAIL PROTECTED] wrote:
On 8/30/2005, Gulzar Hussain [EMAIL PROTECTED]
wrote:
Hi All
I have posted this problem many times on the list
but
no reply
Hi All
I have posted this problem many times on the list but
no reply, trying one more time may be someone will
response this time
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in more
than
Hi
I am using a Lucent MAX TNT to terminate 11 PRIs and
using a single Asterisk box to handle all calls
--- Andrew Thrift [EMAIL PROTECTED] wrote:
We have the ability to do this on a large scale, but
want to do it on a
smaller scale for 1 to maybe a maximum of 5 TNT's.
Andrew Thrift
Hi
no i write this application for my custom needs, but
anybody of you can use it or customized it according
to your needs
cheers
--- Matt Riddell [EMAIL PROTECTED] wrote:
Gulzar Hussain wrote:
Hi All
I just completed a custom application for Asterisk
(i
m not a C guru so i just
Hi All
I want to terminate as much POTS lines as possible to
my Asterisk Server, please advice me which Card to
choose with accessories
Thanks
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
Hi All
I just completed a custom application for Asterisk (i
m not a C guru so i just copy codes from other
application and alter according to my needs)
attached files is the source file
this application is working fine but still i need you
people to give suggestion to improve it
Primary task
yeah i am using chan_zap and i have tried all
combinations of pridialplan and nationalprefix etc.
--- Peter Svensson [EMAIL PROTECTED] wrote:
On Sat, 20 Aug 2005, Gulzar Hussain wrote:
I am having another strnage problem :)
When I dialout on any number from asterisk, it use
to
add
Hi All
I am having another strnage problem :)
When I dialout on any number from asterisk, it use to
add a leading zero in dialed number
for e.g
I dial a number 5832876
and when I check the tracer's result of PSTN switch
that shows me call request for 05832876
thats why I can dial NWD and ISD
Hi
how to disable call waiting on SIP User agents
(incominglimit=1 is Deprecated , End of life already
announced
no idea how to use setgroup to achieve same
functionality)
Thanks
Start your day with Yahoo! - make it
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Hi
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a
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Hi
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a
Hi
I want Queue Application not to call those agents who
are busy talking
is it possible ?
Thanks
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
Hi
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in more
than 90% calls, Caller can hear the voice of the
receiving side
but the receiver cant be able to hear the caller for
exactly 12 seconds,
Hi
I am using asterisk and logging CDR to my SQL Server,
it works fine but if the Connection breaks between
Asterisk and SQL Server it doesnt reconnect itself
does somebody has any patch for doing it
Thanks
Start your day
Hi All
When I call from 1 Windows Messenger to another
without Asterisk everything use to be fine but when
asterisk is there as a Registrar a problem use to
occur in many calls (With Canreinvite = Yes in
SIP.CONF), Caller can hear the voice of the receiving
side but the receiver cant be able
Hi
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in many
calls, Caller can hear the voice of the receiving side
but the receiver doesnt get any voice for
about 5 to 10 seconds, conversation will
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