1- I've tried running fxotune
2- I've tried turning off all un-necessary hardware in the BIOS
3- I've tried on a different PCI slot.
4- I've tried these suggestions:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
5- How I check if it the clicking and popping correlates to
Hi all and thanks for every suggest about my problem, I found that my TDM400P
was sharing IRQ with onboard sound device using cat /proc/interrupts, lspci -v
and lspci -vb. When I disable all unnecessary hardware on my machine and test
it, clicking sounds continue on the line with the same
Hello,
I have a TDM400P with 4 FXO ports, currently using three. When sending or
receiving calls on this card, there is a nearly constant
popping/clicking sound, it is related to the
echo cancellation?. I adjusted my gains properly, but to no avail. I
even found that setting
Hi Everyone...
I am running Asterisk 1.2.22 on Debian Etch. I installed it from
sources. I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded
http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_rxfax.c
Hello all, I post this issue thinking too that could help other people on an
asterisk deployment over distributed offices considering both quality, prices,
devices and so.
Well, i am working on a deployment of a telephony system based in asterisk. My
company have a central office with seven
Steve, Im not asking but looking for a suggest about multiple solutions to the
same problem, Im looking for experinces with hibrid deployments that save me
money, for example sellers offers me TDM04B DIGIUM CARDS about u$s 500 against
u$s 150 for OPENVOX CARDS. Cheers
) in new stack
-- Called g1/38
-- Zap/1-1 answered SIP/ggonzalez-081d13f0
Here Dial cmd do one ring and nothing more, Zap channel has answered but the
number dialed never RING, what is wrong? what i have to do get this working
fine?. Thanks for any help
Thanks for help me, well, I do all that i see on the wiki page about asterisk
and
nat troubleshooting, because did not work I connected asterisk to a public ip
for testing, but, while I get two sip phones with private ip connected to my
asterisk with public ip, I can setup calls(phones rings) but
Hello all, iam setting up an asterisk box behind NAT to get SIP calls from
outside or internet.
In that eschema i can setup SIP calls but, while from the outside nat people can
hear me, Im unable
to listen anything behind NAT. Out of firewalls settings( I checked this to port
fowarding) what can
Iam working on a PANASONIC KX-TD1232 and what i need is a sample of config files
to link this system with Asterisk through the Panasonic VM System. Thanks.
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I have to do this configuration with a panasonic KX-TD1232 model. You need some
other information about the panasonic system?.Thanks.
G.
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Thanks for help me with this issue. I've this scenario, a PANASONIC KX domain
and an ASTERISK domain, each one with their own pool of extensions, incoming
calls are recived by the PANASONIC KX as a gateway from PSTN to the office.
Once a call is recived by the PANASONIC,it bridge the call to
If Someone did that, How I connect extensions.conf with this type of Hybrid
system to work with asterisk inside this schema:
PSTN---PANASONIC KX -- Asterisk
|
|-send internal call
Thanks.
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Hello, well, I need to configure two asterisk box like SIP trunks to send sip
calls from one asterisk to the other and visceversa. So How I setup config
files to get this working?.Thanks.
G.
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I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect
HANGUP from this. Can anyone help me to get it work. Thanks!
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Hello all!
I dont know how affect this issue (jitter buffer) on a SIP implementation with
a
VOIP trunk and I want to know how to setup this item to get a good IP quality
calls without voice delay. thanks for any help.
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Hello all!,
Ive install all package of Vicidial and astguiclient as I read
on a scratch install notes in a CentOs 4 with trixbox. But when I use some AGIS
added in a dialplan of the install documentation i get some sintax error on this
scripts like
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