i folk
warning from diegoviola
from paragway
he say working for bridgecomm
then teliax
then flowroute
isn’t that a lie only?
he need free morocco mobile traffic from me
i refuse him
i say to him if you help me with some web developement i can provide you lowered rate because was a friend of mine
but now i am avoiding him step by step
the asterisk folk may allready know him
190.23.0.0/16
he have a VPS in germany and US
tacking contact from me and from the irc channels to sell traffic to?
WTF
that’s not a traffic but a fad full route
>:@
diegoviola is the VoIp world killer and trouble maker

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We have a tenant who has been having issues with a congested connection and in trouble shooting it we’ve noticed that there seems to be a lot of SIP traffic even when none of the phones are doing anything.

We’ve determined that this traffic is mostly INFO packets generated by setting qualify=2000. I understand that 2000 ms is the default value for the qualification parameter but what I’m unclear on is exactly what the purpose of having asterisk qualify the phones is.

I know that in a NAT situation, qualifications can help keep UDP sessions open in the firewall but in our case most phones are not behind NAT.

I realize qualifying phones is also how asterisk keeps track of who is available for things like BLF but surely it doesn’t need to do that every 2 seconds to keep the BLFs reasonably current.

So I guess my question is what is the real purpose of the qualify setting in a non-NAT situation and can one safely set the qualification as something higher. I’d think something like 15 seconds would be more than enough for BLFs and the like.

Chris

Hello,

I am in bit confusion as I am not able to find point of trouble.
All extensions are configured same way. All are registered, have same
context, voicemail context.
There are around 112 extensions. So I am giving example of 2 extensions. one
going to voicemail fine and other not.

==================================================================================
CLI output: //WORKING

Hi Everyone,

Wondering if any of you folks ever had trouble using *Thomson
TG784
*DSL/Wireless Router/FXS ATA combo? I am looking to use this to connect
users from home to a hosted Asterisk PBX.

Any and all inputs are appreciated.

Thanks

We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI)
cards.

I am having trouble completing faxes. Carrier send calls to me using SIP.
Any recommendation to have some success with Fax.

We trying using T.38 pass through and using G711U codec.

Asterisk Version 1.6.1.1

Thanks,

Dave