Hi,
I have a problem with our office PBX where outgoing FXO Zap channels
get bridged and
i cannot receive or make any phonecalls.
First I disabled flash function and we are using # sign to do
transfers between internal lines
but it still happends from time to time.
So is there a way to specify
On 11/22/05, Cory Andrews [EMAIL PROTECTED] wrote:
Have an application where Cisco phones are being used in a noisy
environmentlooking for some type of external ringer or amplifier so
users can hear the phones ringing over the background noise. Anyone
familiar with such a device?
What
On 9/27/05, Matt Florell [EMAIL PROTECTED] wrote:
We have finished our tests of the new Digium firmware on the quad T1
cards(TE405P/TE410P). Overall it is a big improvement over the version 1
firmware.
Thanks Matt ! Very valuable reading !
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On 9/21/05, Matt Florell [EMAIL PROTECTED] wrote:
We have sevaral call centers as well, and we just restrict a single server
to 50 recordings at once and then we would pass the next recording as an
IAX2 channel to another recording server. It's a scalable system for us that
is relatively
On 8/11/05, Jarek Jarzebowski [EMAIL PROTECTED] wrote:
Hi all,
I am interested in your opinions about using more then one Tormenta 2
card on asterisk server based on Debian - but distribution does not
matter in this case I suppose.
Its not recommend setup, especially when you need to have
On 7/26/05, Walid Azab [EMAIL PROTECTED] wrote:
Anyone tried reverting to SKINNY from SIP. I have a problem I cannot fix and
need to get back to SCCP to be able to use the phone.
I managed to do that yesterday from SIP 7.4 to SCCP 7.2
regards
m.
Unless you gave g729 licence for asterisk i'd suggest to go with some 4 port
gateway with g723 protocol at both ends.
IMHO you would benefit more from G723, VAD and jitterbuffer on those
gateways then from plain iax trunking.
If you want to go SIP you might have to use other port then 5060 which
chan_sccp2 is pretty usable as far as i can tell. So you can take advantage
of full feature set of cisco's and asterisk
On 7/24/05, Andrew Latham [EMAIL PROTECTED] wrote:
Keep in mind that Cisco Phones with the SIP firmware do not support
XML Pushing in theire native format. Using the HTTP
yes
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Lucent MAX TNT ?
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Write some more about your project,
What it suppose to do ?
On 7/3/05, Manuel Soto [EMAIL PROTECTED] wrote:
I'm evaluating a VRU project which has huge requirements. I'm
looking
for metrics but I haven't found anything that cover my requirements
regards
m.
What card do you have ? Is there are jumper setup that you can specify
E1 or T1 ?
E1 cards a shipped set up as T1 by default
regards
m.
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With that dial option you'd have like a cut off in conversation
whenever the file is played.
i think the purpose is to have it in background so you can have normal
conversation.
Also you might wan't to remove everything that is above 4kHz cause it
want affect conversation either way.
rgrds
m.
On 6/2/05, Andrew Latham [EMAIL PROTECTED] wrote:
I don't know, but pricing it per line whould be safe. Say $100 per
line that would be $67,200.00. So anything less than that would be
great. I think it will be about $20 bucks a port.
672 * 20 = 13,400
come on it must be cheaper ! for that
Your explanation is really messy but from what I understand it seem
like you just want to be able to take incoming call from one asterisk server
and forward it to another asterisk sever over IP that would terminate the call
but dont connect calls untill B party answers the phone on second
asterisk
On 5/12/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Digium cards do not have a built in DSP. Neither do the Sangoma as far
as I know. I don't know about VoiceTronix.
As yet !
As for digium cards latest cvs commits suggest that there is some
ongoing development on hardware based
On 5/9/05, Tim Litwiller [EMAIL PROTECTED] wrote:
The Users expierience should go something like this
One of his customers calls in the IVR asks for his account number and
password - based on that it does database lookups for several reports
and reads those back to the customer with text to
On 5/8/05, Steve Rawlings [EMAIL PROTECTED] wrote:
Thanks for that, I can see what you say. But there are two DDI's involved,
one where calls for this client come in for us to deal with which should
pass straight through * to our switch (which is easy) and a second on which
the client can
On 5/6/05, Steve Rawlings [EMAIL PROTECTED] wrote:
I'm using a TE405p with all four spans enabled, two configured as pri_cpe
and two as pri_net, the asterisk is sitting between our ISDN (UK BT
EuroISDN30) and our phone system. We have 200 DDI numbers on the ISDN's and
I need to give one of
On 5/3/05, Michael Bielicki [EMAIL PROTECTED] wrote:
chek your pridialplan setting a well as the prilocaldialplan setting.
Also if you set it as dynamic be aware that it actually strips off the
international and national prefixes, and only tag numbers as
international or national !
same thing here
i bet somone was marking messages from asterisk-users as spam on
constan basis and that's why it's happening
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To
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle.
Anybody had his hands on this card or knows some details ?
regards
m.
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On Tue, 29 Mar 2005 09:20:40 + (UTC), Tony Mountifield wrote:
I have a potential client that wants to send many faxes simultaneously,
over E1 trunks.
How CPU intensive is spandsp's txfax? How many concurrent faxes could
be sent by a decent CPU (e.g. Xeon 3GHz) before timing starts to get
Sure u can.
Buy nokia phone, buy nokia serial cable, use gnokii software to trigger script
on sms received. Script would write sample.call file to asterisk queue
directory.
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There is nice script that you can use cvs2cl.pl
that creates changelog file from cvs entries which has the tags you are
talking about
example
http://asterisk.gnuinter.net/files/changelogs/asterisk-ng.ChangeLog
On Wed, 23 Mar 2005 22:54:35 +0100, David Hajek
[EMAIL PROTECTED] wrote:
Hello,
On Thu, 17 Mar 2005 14:53:27 +0800, XinTai Wang
[EMAIL PROTECTED] wrote:
who have been fabricated their own cards from Tormenta 2 PCI Card?
govarion.com
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you need to get openssl-dev package too
for most dependencies problems you need respecitive dev libriaries
regards
m.
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set asterisk to log into database directly via there are mysql ,
postgresql and odbc drivers
available.
You dont need radius at all,
for billing and accounting all u need is a frontend to database
On Thu, 17 Mar 2005 12:29:34 -0500, Matt wrote:
Oh this is sad.. I'm familiar with radius.. and
If you have some money to spend buy used aurora isdn tester they are
really handy and
they can do lots of stuff including access to voice channels.
Home grown will never be as good as ready to go equipment IMHO.
regards
m.
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I'm not planning to use it but still this is really cool !!!
Thanks
regards
m.
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I agree with peter stick to digium cards at least you'll have native
support for asterisk.
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Hi,
Does anybody has experience with high capacity PSTN voicemail and
asterisk, running more then 5k mailboxes for PSTN users ?
How many mailboxes can I serve with 4xE1 card if we assume that we
have enough harddrive
capacity. What would be server requirements. Would the CPU load be the
same when
On Sat, 29 Jan 2005 11:13:36 -0500, Jim Van Meggelen wrote:
does anyone out there made some experience with Varion
(www.govarion.com) based E1/T1 cards ?
Their cards work. The only problem about govarion is their delivery
time. The cards are just not shipped as promised. And it's not only
On Tue, 25 Jan 2005 23:35:47 -0600, Fernando Romo wrote:
Here in Mexico the R2MF is a standard for the dominat phone company
(Telmex), The Sangoma People tell me the products must work with the R2
librarys wrote by Steve Underwood.
Should work exactly the same as digium cards, i think it uses
Make sure that outgoing calls have different pattern of taking
channels then incoming calls.
If your carrier uses channels in order 1,2,3,4 use outgoing calls like 30,29,28
look at G,g switches for dial application.
This error may be caused by the fact that your telco's swich thinks that
channel
On Mon, 24 Jan 2005 11:38:52 -0500, Jon Bebeau wrote:
I'm exactly in the middle of benchmarking the A104 and T410p. I'm
developing a matrix of CPU, bandwidth throughput and trying to find high
water marks under several loads; single processor, multi processor, Xeon vs.
P4, Hyperthreading vs.
On Mon, 24 Jan 2005 23:21:49 -0500, jbebeau [EMAIL PROTECTED] wrote:
No, I don't work for Sangoma or any affiliate. I receive no compensation or
benefit from Sangoma or any affiliate in any way. I'm just trying to fine
the best stuff for my use.
Great ! I'm can't wait to see results !
best
On Fri, 14 Jan 2005 10:01:52 -0600, Matthew Boehm wrote:
Why does it have to be commercially licenced?
Without it, the SS7 software would be linking to GPL software which
means they would have
to GPL the code too. So the only way to get commercial SS7 is to have
it with commercial
asterisk.
I have following setup
Asterisk - T100P - Adtran TSU600 P + FXOcard - PSTN line
When PSTN line is plugged directly in to analog X100P caller id is
received by Asterisk
but when I plug it into adtran I'm not getting caller id.
Any ideas what kind of setup Adtran TSU600 requires to pass caller
Check out used Lucent MaxTNT equipment there was a thread on the
mailling list about it.
On Tue, 14 Dec 2004 13:14:43 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
And T1s too. If you can supply a seperate piece of hardware that can handle
all the T1 crap then pass calls to asterisk as
Hello,
I'd to get a sample to make sure that card is working could you quote
me on shipping to poland.
It it a full clone ie. will same driver work for your clone card ?
Thanks in advance
regards
m.
On Tue, 09 Nov 2004 18:20:25 -0300, Richard Moore
[EMAIL PROTECTED] wrote:
Hello list ,
I´m
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