quintum gateway server, the status
appears as soon as the call get connected.
cheers
Aby Azid
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Thank you for replying,
How would i know, whether i have the valid indicitions.conf ?
On Sun, Apr 20, 2008 at 8:47 PM, Eric Wieling [EMAIL PROTECTED] wrote:
Make sure you have a valid /etc/asterisk/indications.conf
aby azid wrote:
Hi,
this is my first ever post, would appreciate
Hi Eric,
i copy the indications.conf.sample from the asterisk source and paste it in
the /etc/asterisk directory. I reloaded asterisk and still the message
appear when i sent call to Quintum. Am I doing it right?
cheers,
Aby Azid
On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling [EMAIL PROTECTED
gateway. In this case, I'm using Wellsip from Welltech.
cheers,
Aby Azid
On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling [EMAIL PROTECTED] wrote:
Use the indications.conf.sample that comes with the Asterisk source.
aby azid wrote:
Thank you for replying,
How would i know, whether i have
Hi,
I have softphone with a g723 codec, my question is how do i set it as Pass
thru in Asterisk?
cheers,
Aby Azid
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told me to
use pass-through.
cheers,
Aby Azid
On Thu, Apr 24, 2008 at 10:42 PM, Anthony Francis [EMAIL PROTECTED]
wrote:
More importantly, for it to pass-through you need something that
processes g723 on the other end. If Asterisk is terminating the call by
handing it off to the PSTN
hi,
yes, apparently the status message appeared due to the codec setting from
the gateway.
cheers,
Aby Azid
On Thu, May 1, 2008 at 5:34 PM, Cyril SCETBON [EMAIL PROTECTED] wrote:
Hi,
Did you resolve the issue you were hitting ?
aby azid wrote:
hi,
I hope anyone can tell me how
samples 160
[May 9 12:53:39] WARNING[3626]: indications.c:149 playtones_generator:
Can't generate that much data!
[May 9 12:53:39] WARNING[3626]: translate.c:211 framein: zapg729toulaw did
not update samples 160
*Thank you in advance,
Regards,
Aby Azid
Hie,
I managed to connect two Asterisk box via SIP. My problem is when I login
using Realtime SIP, I will get
chan_sip.c:8373 check_auth: username mismatch, have 8003000777, digest
has voip3
Failed to authenticate user 8003000777
sip:[EMAIL PROTECTED][EMAIL PROTECTED]
;tag=as4d11916d
when
Hi,
I have question regarding Asterisk Local channel. Is it possible to define
codec used in Local channel as like in SIP channel?. If it's possible, how
do i do it?
Thank you
Regards,
Aby Azid
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Hi,
I have question regarding Asterisk Local channel. Is it possible to define
codec used in Local channel as like in SIP channel?. If it's possible, how
do i do it?
Thank you
Regards,
Aby Azid
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or tools for me
to achieve this.
Cheers,
Aby Azid
Vyke Asia
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on the method you used.
Cheers,
Aby Azid
Vyke Asia
On Fri, Aug 15, 2008 at 1:45 PM, Saul Bejarano [EMAIL PROTECTED] wrote:
Remember the rule of 30Mhz per call when you kill the machine and also
the bandwidth usage on connected calls.
Kind regards,
Saul Bejarano
aby azid wrote:
Hi everyone
. Is there such
tools and ways for me to create simultaneous AIX2 calls?. Again, really
appreciate if anyone can come up with ideas or tools for me to achieve this.
Thank you in advance,
Regards,
Aby Azid
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