Why don´t create tariffs on your a2billing with full numbers of your
SIP users? You can point it to a trunk that dials sip/%diallingnumber%
--
Antonio José dos Santos Brandão
On 6/20/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
Is there any way to write a custom context, where
Andre,
A better is aproach to R2 in brazil is to use digivoice board (instead of
digium's). The card has native support to R2 in the channel driver.
The best would be to change to ISDN. Have you already asked your telco?
--
Antonio J. S. Brandão
On 3/19/07, andreunimed [EMAIL PROTECTED]
Testing 1.4 here i got the same issue.
Running tcpdump figure out that packets are sent from the sip provider
or ATA to asterisk1.4 machine but asterisk doesn't reply. At really,
nothing apears at /var/log/asterisk/full and looks like the sockets
aren't open.
After a stop now and restart all
Ricardo, I'm looking for the same thing. Have you tried the patch? Got
any success?
--
Antonio J. S. Brandão
On 7/7/06, Ricardo Martins [EMAIL PROTECTED] wrote:
Hi all. I´m trying to disable this simple thing: I dont want an user to
put a call in hold pressing hook (or flash button). I tryied
FYI, just upgraded from 1.2.2 to 1.2.3 and audio problems in sip
channels gone way.
Thanks a lot,
--
Antonio José dos Santos Brandão
Virgos Tecnologia da Informação
www.virgos.com.br - São Carlos,SP
On 1/25/06, Olle E Johansson [EMAIL PROTECTED] wrote:
This morning we discovered a serious bug
.
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Antonio José dos Santos Brandão
Virgos Tecnologia da Informação
www.virgos.com.br - São Carlos,SP
On 9/24/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Has anyone on-list tried using a USB style adapter like the VTA-1000 to
provide some form of gateway from Skype into *? If so, how well
slot. Is there another way to balance IRQs? There is IRQ numbers
unnused...
I've done tests with many codecs. The network bandwidth is ok.
tks a lot. Any comment is welcome.
--
Antonio José dos Santos Brandão
Virgos Tecnologia da Informação
www.virgos.com.br - FWD
