Roberto
Check your router if ALG or similar feature is enabled. Disable and test.
Also, on SNGREP check if both parties are getting ACK correctly after RTP
starts.
*--*
*Atenciosamente,*
*Luciano Moreira**(85)99974-2750*
*__Logic Telecom*
*0800-085-7799 | (85)4042-7799 | **(11)4210-7799
.html
__
Luciano Moreira
Logic Telecom LTDa
Fortaleza, CE
+55 (85) 4062-9150
+55 (85) 9701-2444
+1 360-717-1506 (USA)
2010/6/22 Tarek Sawah :
> i have been struggling with call center Customers for a couple of years
> now.. i have a call center with 40 agents using elastix.. and quality is
>
List members,
When I dial to a PSTN number, the A2Billing script does all the tasks,
until it shutdown without make the dailout by sip trunk set.
Lasts outputs fro the a2billing.php debug are:
a2billing.php|2: RESFINDRATE::> 0
a2billing.php|2: UPDATE cc_card SET inuse=inuse-1 WHERE username='5
I trying to setup a outbound trunk with IPSmarx. It's working, but when I make
a call, the ring dialtone stills ringing on my side, even after the other side
picksup the phone. I got a NOTICE message from Asterisk that I hope you can
help me:
-- Called [EMAIL PROTECTED]
Caros,
I installed the A2Billing - v1.2.2 with Asterisk 1.2.10. All works ok, but when
I try callout got a message saying the number in not available.
Can you help with a step-by-step to make a card autenticate and dial a number?
Thank you
Luc Moreira
Mais VoIP
-- Accepting AUTHENTICATED
the
call stays mute.
any clues about this bug?
Luc
---
Luciano Moreira
Sip Phone #:
1-747-661-7629
Phone #:
+55 (85) 3263-0372
+55 (85) 9956-2956
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Pessoal,
Facing wierd bug on * using MD3200 modem. It was working ok, then
after a boot bug started:
The calls that came out thru the Zap channel, stopped work. The *
gets the call from an IAX client and set it as active, even before
the destination rings, and finaly when someone pickup de phone
