Hi all,

Sorry for posting here – but I figured there are many people with Cisco
IP phones here – and I use them with Asterisk

I have a couple of Cisco 7940 phones. I’ve loaded the SIP firmware OK,
loaded the SIP configuration files OK, they work with Asterisk just fine.

My question is – will I have to keep on running the tftp server for them
for ever and ever? Isn’t there any option for them to just use the
settings they have already loaded form the tftp server – so that I can
kill tftpd on my server machine? I tried doing that, and then the phones
stop booting, going in a loop looking for the tftpd server.

It seems a bit pointless, having to run the tftpd daemon all the time -
although I’ve already loaded the firmware and configurations I want.

Thank you,

Sebastian

On Sat, May 07, 2011 at 04:24:08PM +0100, –[ UxBoD ]– wrote:
> I know a lot has changed over the past couple of years, and even monthly, and that Asterisk running within a virtualised environment is very happy indeed. If one would only be using SIP/IAX would Xen/KVM be the best solution ? / or perhaps VServer/LXC maybe advantageous due to binary hashing. Your thoughts would be very welcome.

VServer / OpenVZ / LXC run the the instances on top of the same kernel.
They use kernel-level seperation. So this means that they basically see
the subsets of the same file system and such.

KVM / Xen provide the view of a different machine. The guest machine run
its own kernel. I guess this is better if you don’t trust the guest, but
you pay more in resource consumption. If you want to run a different
kernel, you must use this one.

Hi All,

I have set following milliwatt and dial in from my mobile phone. and i adjust txgain value in chan_dahdi.conf but no effect on dahdi_monitor value is same. even i change its to 20 and more but monitor still showing same value. But i can hear tone variation on my mobile when i am changing txgain… is there any specific value i can set in txgain or leave it default 0.0

( # = Audio Level * = Max Audio Hit )
< ----------------(RX)----------------> < ----------------(TX)---------------->
#########################* 0) Tx: 4705 ( 4719)

[milliwatt]
;Asterisk rx/tx gain testing
exten => s,1,Answer
exten => s,n,PlayTones(1004/1000)
exten => s,n,Wait(300)

Hi,

I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example

SIP—————–>PRI ————> mobile

I have set progress=yes in chan_dahdi.conf but still not working

if i call inbound from my mobile to internal extension ringing working

please help me

-S