Hi,
I'm want to do something slighty different with call queues than the
config allows...
I wish to have things work in an 'overflowish' manner. Ie, it works just
like 'roundrobin', where it rings
on one phone, no answer, rings on the next etc etc, except I want it to
keep ringing on all
Since when does DHCP enforce an IP address?
You can configure it to always give the same IP to a particular MAC,
however this dosn't stop a malicious user with 1/10th of a clue :D
As Gavin Hamill suggested, iptables rules would be perfect to do things
like 'only accept traffic from this IP
Any ideas what bugs are fixed?
there dosn't seem to be any release notes.
-Shaun
Rod Bacon wrote:
The firmware at...
http://www.snom.com/download/share/snom190-3.60b-SIP-j.bin
seems to have fixed quite a few SNOM 190 bugs. Worth a try.
___
Hi,
I think the correct answer to this question is whatever you are most
comfortable with.
I also use Debian Sarge and find it to be great. I'm also a FreeBSD user.
Perhaps the best thing to do would be to install one distro and have a
play and see if you like it.
If you don't, try something
Hi All,
I'm hoping that some one has come across this one before or knows of a
better solution for this problem...
I've (partially) written a perl AGI script with Asterisk::AGI. I need to
be able to execute a macro within
the dialplan and pass it a single option (the external phone number to
Scott Bussinger wrote:
We just tried to go entirely with softphones in our office gave up after a
month or so of trying. I tried probably 10 different softphones running on
3.0Ghz WinXP machines and none of them were workable. I tried both SIP and
IAX2 softphones using headsets plugged into the
is beyond me. The GUI mixers all list
this channel simply as WAVE, and I have two WAVE channels. So remember
amixer.
The card is called ICEnsemble ICE1230 by OSS, and Yamaha DS-XG (YMF734)
by ALSA
On Tue, 2005-22-03 at 10:52 +0800, Shaun Dwyer wrote:
Scott Bussinger wrote:
We just tried to go
Hi Josh,
Thanks for the info..
how did you get intercom=true into the URI, and onto the end of the
INVITE line?
btw, I got an intresting response from Sven of Snom...
[Sven Fischer (support) wrote:]
Hi,
as far as I understood intercom will only work if you are not using any
password for
about getting this working,
please do email me :)
Cheers,
-Shaun
Shaun Dwyer wrote:
Hi Josh,
Thanks for the info..
how did you get intercom=true into the URI, and onto the end of the
INVITE line?
btw, I got an intresting response from Sven of Snom...
[Sven Fischer (support) wrote:]
Hi,
as far
Hi All...
I'm trying to get the intercom feature working on some snom 190 phones
but having no luck...
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended
to the To: header as per requirements. I've email'd snom a few days ago
but have yet to
get a
Or open up a firewall rule to allow access to the IP that asterisk is on
from the otherwise isolated
subnet.
You could also make the asterisk box 'multi-homed', ie, put a 2nd nic in
it and plug it into
both subnets, then in sip.conf set the bindip to 0.0.0.0 (all interfaces).
In essence
Hi Altus,
I believe the only reason you require a Zaptel card is for a timing
source for trunking.
You could probably get away with the ztdummy device that uses USB as a
timing source.
Go have a look on www.voip-info.org for more info on ztdummy.
Cheers,
-Shaun
Altus Snyman wrote:
Good day all
also read
Snom's mass deployment
documentation but thats no real help in this case.
Cheers,
-Shaun
Original Message
Subject:Snom 190 function keys via subscribed config
Date: Tue, 08 Mar 2005 11:15:18 +0800
From: Shaun Dwyer [EMAIL PROTECTED]
To: [EMAIL PROTECTED
/
To see how it wants the settings, manually configure a phone and look
what the Settings
tab at the bottom of the bottom of the Lefthand side menu shows...
Later;
Tim
On Mar 9, 2005, at 9:35 PM, Shaun Dwyer wrote:
Hi All,
I realise this is off topic, but its likely the best place to ask!
I sent
, at 9:35 PM, Shaun Dwyer wrote:
Hi All,
I realise this is off topic, but its likely the best place to ask!
I sent an email to snom support a few days ago but have yet to
recieve a response..
Perhaps some one has found a solution to this problem already? I've
searched
the mailing lists and google
Hi,
You may want to look into LVS (Linux Virtual Server). It allows load
ballancing in a highly configurable way.
http://www.linuxvirtualserver.org/
We use it on our web and mail server to load ballance across multiple
hosts. The way we have it configured
it will maintain a session for 15
Hi,
I'm looking at connecting an analog fax to asterisk via an FXO card.
The plan is to send faxes thru freshtel.
Has anyone done faxing with freshtel?
Cheers,
-Shaun
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Has anyone seen/heard/has this phone working with asterisk?
I've done some searching and found that there is zero information about
this phone. Probably because it dosn't work.
One of the Wholesalers I can get stuff from has them going quite cheap.
Probably because he couldn't sell them :)
Cheers,
Hi,
I'm having some problems getting calls to go out via freshtel.
There dosn't seem to be any specific information on how to get it
working anywhere.
The only information I've found is here:
http://www.voip-info.org/wiki-Freshtel
and that dosn't give you any idea of how to actually get it
Shaun Dwyer wrote:
Hi,
I'm having some problems getting calls to go out via freshtel.
There dosn't seem to be any specific information on how to get it
working anywhere.
The only information I've found is here:
http://www.voip-info.org/wiki-Freshtel
and that dosn't give you any idea of how to actually
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