Simon Coles wrote:
--On Tuesday, March 2, 2004 9:49 am + Steve Kennedy
<[EMAIL PROTECTED]> wrote:
That's the crunch (1.5/512) ... it's actually the 512 which is relevent.
Virtually all DSL in the UK is a wholesale product from BT (they have
about 2 million customers, Easynet who local loop
Did that, got the same result. Any other ideas? Do I need a STUN server or
does * have something built in for dealing with the RTP packets?
Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
Office: (217) 585-0357
Cell:
Hi,
I would like the Asterisk SIP channel to be able to handle 60ms of voice
per packet instead of the default 20ms. It would be very nice to have
this as a config option or at least to be able to hardcode it in the
source files. Most SIP gateways allow you to do this and it would be
great i
Greate, I was thought I had done something to my installation, I cant use
iaxtel's 1-8XX numbers.
So they might be down.
-- Executing Dial("SIP/hha2-bf35",
"IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 69.73.
G.729 is just about everywhere. A lot of boxes use G.723 (and/or G.726) too
but G.729 ends up with about the same quality but at a much lower bit rate.
Most inexpensive "hard" phones don't use G.723 because it takes a lot of CPU
power. G.729 gets better results in the same or less CPU. We always
I agree but I have never seen it actually implemented in anything other than
a softphone. If you know of any "hard" phone out there that supports it
please let me know.
Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
I looked all over google and the mailing lists but I can't figure this out.
I can call a non NAT to a non NAT without a problem with and without
reinvite. As soon as I try to have a call between a NAT and a non NAT I get
this... The phones can't hear each other. The MikeNATSnom1 has nat=yes and
On Mon, 8 Mar 2004 23:45:44 -0500, Andrew Kohlsmith wrote
> > G.729 is the best codec IMHO. Goto
> > http://www.packetizer.com/iptel/bandcalc.html to get the real scoop.
>
> IMO ILBC beats G.729 in terms of quality, especially in environments
> where packet loss can occur. It's about the same c
> G.729 is the best codec IMHO. Goto
> http://www.packetizer.com/iptel/bandcalc.html to get the real scoop.
IMO ILBC beats G.729 in terms of quality, especially in environments where
packet loss can occur. It's about the same computationally-speaking but
it's also royalty-free.
GSM is my seco
G.729 is the best codec IMHO. Goto
http://www.packetizer.com/iptel/bandcalc.html to get the real scoop.
Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E
http://www.buoy.com/mailman/listinfo/monastery is where you can sign up
and join the chatter about Monastery. Support and announcements will be
on this list
Tim
--
><
>> Tim Sailer >< Coastal Internet
On Mon, Mar 08, 2004 at 10:06:22PM -0300, Bartosz Jozwiak wrote:
> Hello,
>
> For I would like to say that this is great tool!!!
Thanks!
> I have one question. Why my busy indication
> is turned on even when person hangup a call ?
IAX or SIP?
> I have noticed that this happens only to phones
>
So put your hands on it and help to product grow.
Regards,
Gus
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 08, 2004 8:19 PM
Subject: Re: [Asterisk-Users] SIP - Receptionist
> Monastery is neat as a monitoring tool. The console's we're
> t
Thanks for the info so far. I am still trying to asterisk'ize my ML9.2
firewall box and can't get the external SIP registration to work. If I
hook up my Sipura directly to the WAN it registers OK.
This is the message I get from asterisk:
Mar 8 21:03:07 NOTICE[196621]: chan_sip.c:3140 sip_reg_t
I am just starting to experiment with Asterisk. I know that almost all
IP Phones and devices support either G.711 (alaw/ulaw) as codecs, but these do
not really provide any bandwidth savings when used throught the internet or
private links. I have used GSM with some software phones and I lik
On Monday 08 March 2004 07:31 pm, Brian Capouch wrote:
> Well, maybe. The Grandstreams use an extended version, and at least
> right now, the hpa tftpd does NOT work with them. I spent many hours
> playing around last night, and the GS phones for whatever reason will
> not download files that ar
Hello,
For I would like to say that this is great tool!!!
I have one question. Why my busy indication
is turned on even when person hangup a call ?
I have noticed that this happens only to phones
extension where a lot of incomming call trafic is
generated. Is it something to do with call waiting?
Hello,
I am testing "*" in a developer computer.
Can I uninstall the license and transfer it to a production computer in
future ?
[]s
Marcio Gomes
Derek Samford wrote:
As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what
I have your same setup: Asterisk running on a box that
also runs SAhorwall. I can register to both WD and ICH.
One thing I suggest is first getting Asterisk to work
without shorewall.
Next install the firewall but leave it wide open, close it
down incrementally. Also turn on logging of every d
Hi,
Has anyone out there had any luck with channel return codes with
chan_h323? It seems that the h323 return codes are in the channel
driver for for h.323 debug messages, but for some reason, there is no
distinction between busy and congestion returning to Asterisk, so it's
not possible to te
Hello all,
I'm looking for advice for codec that works best
for asterisk. Anyone has real testing with all codecs, specially with
G.729. I have tested with single call on few codecs that come with
asterisk by using IPTraf and the rate as of below:
ulaw 64 Kbps, sample-based Also known a
[EMAIL PROTECTED] wrote:
Hi All!
I am thinking about fork-lift-upgrading a Nortel-Meridian
key system with a * PBX driving SIP phones in the office.
The interface to PSTN would be a fractional T1 PRI (11 lines
plus D channel). The GS phones look acceptable for most
users. The forthcoming "Sayson
Sweet will continue this on the other list...
- Original Message -
From: "Michael Van Donselaar" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 09, 2004 1:08 PM
Subject: Re: [Asterisk-Users] DIAX Error
> On Tue, 9 Mar 2004 12:35:40 +1300, "Matt Riddell" <[EMAIL PROTECTE
On Tue, 9 Mar 2004 12:35:40 +1300, "Matt Riddell" <[EMAIL PROTECTED]> wrote:
>Also,
>
>If I click a phonebook entry then dial it crashes...
what are the contents of the phone book entry?
>
>The reason is that it goes to 2 for line and that doesn't exist
?
>
>If I click VM then 1 then dial, it
>
> IAXPhone doesn't run on 98
>
> :-(
>
Ok. I rebuilt the installer to accept Win98. Please give it a try when you
get a chance. http://www.sokol-associates.com/
Thanks,
Steve
> Strange, I have no problem with echo!
>
> :-)
>
> Matt
> - Original Message -
> From: "Steven Sokol
This is my first post to the list, and while I am sorry that
I have a problem that I need to bring to the list, I have
been a very satisfied reader/lurker on the list, which has
saved me from asking lots of questions so far :-).
Apologies in advance for the length...
I am new to *, but am already
Thanks for your advice also Colin, I would like to stay away from POTS
cards in the server but it is an option if all else fails.
Philipp von Klitzing wrote:
Hi!
What I want to look at having a Small Home Office setup were I can use
the 1 BRI for both DID and Internet at the same time.
Is it
Hi!
> Initial thoughts are to use a counter, increment on call presentation,
> decrement on call tear down, and give the inbound call busy or congestion
> treatment if the counter is above a certain value when the call is
> presented?
I guess you need to "protect" your rather thin Internet uplink
Also,
If I click a phonebook entry then dial it crashes...
The reason is that it goes to 2 for line and that doesn't exist
If I click VM then 1 then dial, it's ok
Matt
P.S. is there somewhere I sould post this instead of here?
---
Outgoing mail is certified Virus Free.
Checked by AVG ant
is there any documentation on porting the asterisk program to windows? if
not where can I get info on how to see if I can do this? thanks
hank
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 08, 2004 11:07 AM
Subject: Re: [Asteris
Hi ..
When the receptionist parks a call for someone who is in the
building
but not at their desk, she does an 'allpage' which
blares-out over
the intercom system: 'Willy you have a call parked at 101'.
Willy can then just grab any phone (kitchen, hall, computer
room)
and pick up the parked call.
Monastery is neat as a monitoring tool. The console's we're
talking
about also let the user pick-up calls etc.
- Original Message Follows -
> See monastery, maybe help you
> (http://pbx.unslept.com/newstatus.php)
>
> Regards,
>
> Gus
>
> - Original Message -
> From: <[EMAIL P
also...
I cannot turn of the beep!
If I turn it off and go back, it's back on again!
Matt
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.614 / Virus Database: 393 - Release Date: 3/5/04
IAXPhone doesn't run on 98
:-(
Strange, I have no problem with echo!
:-)
Matt
- Original Message -
From: "Steven Sokol" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 09, 2004 12:01 PM
Subject: RE: [Asterisk-Users] DIAX Error
> Matt,
>
> While you're testing these,
Matt,
While you're testing these, could you give IAX Phone a shot. I just want to
see if you run into any serious differences/errors. BTW - Echo Cancellation
does not work in ANY of our impelemntations (DIAX, iaxComm, IAX Phone).
Regards,
Steve
Steven Sokol
Owner/Manager
Sokol & Associates, LL
BTW...turns out that IAXcomm is exactly the same with regards to audio
quality!
Hope I haven't put anyone off this!
I have a problem with network that changes regularly!
Will update...
BTW heaps of bandwidth free!
Matt
- Original Message -
From: "Matt Riddell" <[EMAIL PROTECTED]>
To:
Hi!
> What I want to look at having a Small Home Office setup were I can use
> the 1 BRI for both DID and Internet at the same time.
>
> Is it possible to use something like a FRITZ! ISDN BRI card to have full
> time Internet on one of the B channels and have the other B channel for
> * for bo
Konrad Gorski wrote:
maybe CRC problem?
try:
span=1,1,0,ccs,hdb3,crc4
No, the provider told us no CRC (and I checked anyway, they weren't
kidding).
Nick
___
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[EMAIL PROTECTED]
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Hello,
We've made another release of the astguiclient suite of client GUI
interfaces for Asterisk(please note this is not a config file editor).
This release has a lot of bug fixes and includes the VICIDIAL
one-call-at-a-time dialer. We have also finished our new website complete
with a new swan
--On Tuesday, March 2, 2004 9:49 am + Steve Kennedy
<[EMAIL PROTECTED]> wrote:
That's the crunch (1.5/512) ... it's actually the 512 which is relevent.
Virtually all DSL in the UK is a wholesale product from BT (they have
about 2 million customers, Easynet who local loop unbundle may have
2
I have cvsed to the 3/9/04 version. I have not been able to do trunking
with other iax2 server. I was able to do it before.
What are the procedures to diagnose this problem? Would it be firewall
related? Would it depend on both peers with the same cvs version?
--
David Kwok
Tel: 612 99292086 ex
On Mon, Mar 08, 2004 at 03:12:52PM -0600, [EMAIL PROTECTED] wrote:
> Simon,
> Do the GS phones support stutter tone as-well-as
> the message light?
I'm not Simon, but yes, they do. At least my -100 does. The display
backlight flashes, and you get the stutter dialtone.
Tim
--
>>
On Mon, 2004-03-08 at 18:50, Olle E. Johansson wrote:
> Jean-Marc V. Liotier wrote:
> > On Sun, 2004-03-07 at 13:03, Philipp von Klitzing answered off-list:
> >
> >>>[6040]
> >>>defaultip=192.168.1.40
> >>
> >>Replace this with "host=dynamic" and see what happens.
> >
> > That's it !
> >
> > Thi
> I am struggling getting asterisk to work on my firewall box.
>
> The Linux box is a firewall running Mandrake 9.2 and
> shorewall for security and NAT. Asterisk is compiled and
> running on the firewall box with a modified sample
> configuration. I am connecting to it using a Sipura on the
I've had a quick look over the wiki and played around with my config a bit
but still can't seem to come up with an easy answer to what I want to do..
I have an Asterisk box set up and have a DID from iconnecthere. They allow
multiple simultaneous inbound calls to that number. My question is: Is th
I use Putty.
It runs on windows, is free, and connects to my various servers.
It allows you to get an SSH terminal (just like a normal Linux terminal) and
you can use it to edit your config files as well as running the asterisk
command window.
You will however still need to edit the config files
See monastery, maybe help you (http://pbx.unslept.com/newstatus.php)
Regards,
Gus
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 08, 2004 6:27 PM
Subject: [Asterisk-Users] SIP - Receptionist
> Hi All!
> I am thinking about fork-lift-upgradin
Webmin will allow you to handle your Samba configuration without any need for
touching config files. You can download webmin at www.webmin.com. Its a
really useful Web-based configuration utility for Linux and Sun servers that
allows you to control all sorts of servers and services on your Linux
Have to say, the new DIAX is heaps better!
Calls seem to be going through a lot cleaner (not breaking up as much) and
is almost the same as Firefly in terms of this.
IAXcomm is still unusable in this regard.
Error's from DIAX:
(Running Win98SE with all the latest patches on a machine windows wa
Hm.. I have to agree with Tim - it's intersting how many people are
interested in such a thing.
So, here you go:
http://graphics.cs.uni-sb.de/VoIP/astui.tar.gz
Complete with monastery, MD5 encryptest sip-secrets and sip password
change.
Enjoy,
Rainer
Note: This are just some scripts we'
Hi All!
I am thinking about fork-lift-upgrading a Nortel-Meridian
key system with a * PBX driving SIP phones in the office.
The interface to PSTN would be a fractional T1 PRI (11 lines
plus D channel). The GS phones look acceptable for most
users. The forthcoming "Sayson 480i" would work for
manag
Theoretically, Samba should do it. See samba.org.
Unfortunately, it has not fully worked for me. Linux can see my Windows
files, and Windows can see the Linux box but can not sign on. I've tries
using the Linux sign-on and password, and also make entries in smbuser.conf,
but had no luck. I'm sure
Hi all,
...on to my next issue with Asterisk that I'm trying to get resolved. I have
a Cisco SIP phone (7960), and I'm using Vonage through their Motorola box and
an X100P. All the basics work well.
The problem I'm having is that I'm not sure how to answer call waiting on the
Vonage line fr
hank smith wrote:
hello I am just curious if there is any windows alternitives to Asterisk?
can I also use them with free world dialup?
thanks
hank
I've never used either of these and I'm certainly no authority on the
subject, but here are a couple Windows based alternatives I've come
across whi
Simon,
Do the GS phones support stutter tone as-well-as
the message light?
I am thinking about buying a load of GS-102's
for the office.
Any other comments appreciated.
TIA
Willy
- Original Message Follows -
> Haha
>
> The magic tweak,, I knew there had to be one.
> That works great t
Thanks for the information. You have saved me a few hours on the phone
with TAC.
Low, Adam wrote:
We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now
Probably be easiest to use something like an Eicon Diva T/A which breaks
down a BRI to 2 pots or 1 data and 1 pots or a 128K data- you'd use the Diva
as a bridge to the BRI. It'd sorta look like:
PSTN
| |64K serial-Your PC
|Diva |
| |POTS
Haha
The magic tweak,, I knew there had to be one.
That works great thanks
Simon
> Simon Chappell wrote:
>> Hi al
>>
>> I have 3 GS 101's plugged into asterisk.
>> They work great and teh quality of sound I can not fault. Most people I
>> am
>> speaking to now ask if I have a new phone because
On Monday 08 March 2004 13:59, hank smith wrote:
> is there a program that I can install on my linux box so I can
> configure the pbx from the internet from my windows box so I don't
> have to work with config files?
In a word, no. There are a few GUI applications in the process of
being develope
Hi,
I am struggling getting asterisk to work on my firewall box.
The Linux box is a firewall running Mandrake 9.2 and shorewall for
security and NAT. Asterisk is compiled and running on the firewall box
with a modified sample configuration. I am connecting to it using a
Sipura on the local LAN. T
[EMAIL PROTECTED] wrote:
SSH
>> Nice :)))
On Mon, 8 Mar 2004, hank smith wrote:
is there a program that I can install on my linux box so I can configure the
pbx from the internet from my windows box so I don't have to work with
config files?
thanks
hank
- Original Message -
From: "
On Mon, Mar 08, 2004 at 08:14:00PM -, Simon Chappell wrote:
> Hi al
>
> I have 3 GS 101's plugged into asterisk.
> They work great and teh quality of sound I can not fault. Most people I am
> speaking to now ask if I have a new phone because the quality is so much
> better.
> My latest quandry
Simon Chappell wrote:
Hi al
I have 3 GS 101's plugged into asterisk.
They work great and teh quality of sound I can not fault. Most people I am
speaking to now ask if I have a new phone because the quality is so much
better.
Don't ever use a Cisco phone if you're happy with your GS phones right no
Hi al
I have 3 GS 101's plugged into asterisk.
They work great and teh quality of sound I can not fault. Most people I am
speaking to now ask if I have a new phone because the quality is so much
better.
My latest quandry is to do with the message button and stuttertones. I
dont get either.. If i h
SSH
On Mon, 8 Mar 2004, hank smith wrote:
> is there a program that I can install on my linux box so I can configure the
> pbx from the internet from my windows box so I don't have to work with
> config files?
> thanks
> hank
> - Original Message -
> From: "Steve Underwood" <[EMAIL PROTEC
So where on the net??? ;-)
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Hickman
Sent: Monday, March 08, 2004 11:40 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hotel wake-up
Small world...
is there a program that I can install on my linux box so I can configure the
pbx from the internet from my windows box so I don't have to work with
config files?
thanks
hank
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 08, 2004
David Hickman wrote:
man I am tired. I did not mean for my last email to go out to the whole
list :(
Don't worry about it man; the list is set up to cause that sort of thing
to happen, and if you read it for a while, you'll see that it's an
almost-daily occurrence.
Sort of like a free soap op
On Mon, 2004-03-08 at 19:54, Stephen R. Besch wrote:
> Make sure that you don't have the "Send Flash Event" option set to "YES"
> in the GS configuration. If you do, flash will not be sent as a SIP
> event and the flash button won't work.
That's the secret, many thanks.
I can now get two lines
man I am tired. I did not mean for my last email to go out to the
whole list :(
sorry about that. I just seen alot of people call me via iaxtel.
dhh
-- David Hickman
Pots314-865-4752x1 business x31 home
FWD 23633 """"
IAXTEL
Small world. I have been running * for about 6 months. I really
like it.
I use it to connect my mother in law in arkansas and to integrate into
fwd and iaxtel.
I also have a hidden menu that give me the status of key network
servers and by using x10 a way to kill the power supply of my e
Matthew Marlowe wrote:
You may also obviously check Grandstreams site instead of another
providers site.
http://www.grandstream.com/BETATEST/
A free windows TFTP server is available from solarwinds.net which works
great.
Obviously linux can use tftpd
Well, maybe. The Grandstreams use an extende
- Original Message -
From: "Matthew Marlowe"
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Monday, March 08, 2004 2:02 PM
Subject: RE: [Asterisk-Users] monastery devel page
I'm currently also busy to give it another look'n'feel using just some
CSS. (a screenshot - all extensions blanked
Asterisk needs a timing source to do conference.
If you don't have a Zaptel interface installed but have a uhci-usb interface
you can use the ztdummy to generate the timing.
To use it you need to go to the zaptel driver directory and modify the
Makefile removing the double minus (--) in front of
On Mon, Mar 08, 2004 at 01:02:26PM -0600, Matthew Marlowe wrote:
> Is this ever going to be available?
If you mean the original version from Tim, it still is, see his first
monastery-post.
The url is ftp://buoy.com/pub/asterisk/monastery.tgz
But as he said, he's still working on it.
Cheers,
Title: Queue to zap group
Hi guys,
I looked at the wiki and associated documentation but I'm still not sure about queueing to a zap channel group. In other words I want to put everyone who comes in on MOH unil a line is free on a zap group like g1 at which point it connects it. Any ideas?
hank smith wrote:
hello I am just curious if there is any windows alternitives to Asterisk?
can I also use them with free world dialup?
thanks
hank
No, but maybe you could port Asterisk to Windows. No, that's not a joke.
The Zaptel drivers might be tough, but Asterisk's VoIP features would
proba
Can Asterisk act as a SIP conference bridge? Looking through the source I
notice that it's required to have a Zaptel interface installed. Why is this
a requirement? Can you not mesh the VoIP streams together?
Thanks,
T..S
___
Asterisk-Users mailing
Is this ever going to be available?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rainer
Jochem
Sent: Monday, March 08, 2004 1:11 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] monastery devel page
> not sure how you have the gui setup right now,
Does using registration via ip instead of user/pass provide any better
stability or anything of the liking?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, March 08, 2004 12:50 PM
To: Asterisk Users
Subject: Re: [Asterisk-U
dkwok wrote:
Has anyone using the flash button on GS101 to access call waiting?
My experience is that it does not work. I read in the list that it may
need to tweak the flash duration to under 100msec. Has anyone have any
solution?
Make sure that you don't have the "Send Flash Event" option set
> not sure how you have the gui setup right now, but being in the same boat
> and sucking on the gui front myself, I think what works best is have some
> sort of "tags" a user can put in their own html page, and your app simply
> resolves those. For example using javascript to check for your state
hello I am just curious if there is any windows
alternitives to Asterisk?
can I also use them with free world
dialup?
thanks
hank
Hi all,
is there any reasonably good management tool for Asterisk out there? all
I've found under
http://www.voip-info.org/tiki-index.php?page=Asterisk+GUI are not so
complete utils, as some have the same functionality others do...
Does such "ideal" tool exist or do I have to type ahead all t
Jean-Marc V. Liotier wrote:
On Sun, 2004-03-07 at 13:03, Philipp von Klitzing answered off-list:
[6040]
defaultip=192.168.1.40
Replace this with "host=dynamic" and see what happens.
That's it !
Thinking it was going to make things easier to diagnose, I had chosen to
set the phones with a stati
Anyone know what this means?
Mar 8 12:28:50 NOTICE[-112661]: chan_sip.c:3150 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
Mar 8 12:28:50 WARNING[-112661]: chan_sip.c:455 __sip_xmit: sip_xmit of
0x82ac7b4 (len 366) to 192.246.69.223 returned -1: Bad file d
I anyone able to get calls from IAXtel, I have been trying to call between
to * systems all day with no luck. Worked fine Friday.
Thanks,
Scott
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To
Hi all,
Thanks everyone who replied for their help. I wanted to post and describe my
"final" config in case anyone in the future wants to accomplish something
similar.
Here's my "Standard extension" macro:
exten => s,1,Dial(${ARG2},15)
exten => s,2,Voicemail(u${ARG1})
exten => s,3,Hangup
exte
In CCM add a Gateway. Use H.323 with H.225 as the device protocol.
Next add a route pattern to identify which calls to direct to *.
Lastly use chan_oh323 instead of chan_h323, as the former works
with CCM and the later does not (one way audio).
The setup is extremely easy and works just fine, wi
> On Mon, Mar 08, 2004 at 10:35:54AM -0600, Matthew Marlowe wrote:
>> When is this going to be available?
>
> Sometime this week. I want to make a few other changes before I release
> this version. I need to make it look better. My UI skills suck royally,
> so I want to make it look a bit better, w
Hi
I would like to know what hardware I need to set up my VOIP/PBX system !
I have 4 T1/E1 phone lines.
If I choose Digium for instance !
Which cards do I need to buy to start ?
Thanks
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Asterisk-Users mailing list
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Hey all,
Does IAX do any sort of echo cancellation?
I'm still trying to get faxing working via an IAX peer (from a faxmodem
hooked up to my Sipura -> Asterisk -> IAX Peer -> PSTN Fax), and one of
the things I've read may be necessary is disabling echo cancellation. I've
disabled it on the Sipura,
On Mon, Mar 08, 2004 at 10:35:54AM -0600, Matthew Marlowe wrote:
> When is this going to be available?
Sometime this week. I want to make a few other changes before I release
this version. I need to make it look better. My UI skills suck royally,
so I want to make it look a bit better, which will
On Sun, 2004-03-07 at 13:03, Philipp von Klitzing answered off-list:
>
> > [6040]
> > defaultip=192.168.1.40
>
> Replace this with "host=dynamic" and see what happens.
That's it !
Thinking it was going to make things easier to diagnose, I had chosen to
set the phones with a static IP. Apparentl
When is this going to be available?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: Monday, March 08, 2004 10:56 AM
To: Asterisk Users
Subject: [Asterisk-Users] monastery devel page
I have had quite a number of people asking if I would i
Our telco here in Bulgaria has this option - you can get what number of
lines you want on a primary (E1 here).
You choose how much lines - from 1 to 30 - go to the primary. Price is
according to number of channels, but signalling is primary!
Check out at your telco if this is not possible out th
[EMAIL PROTECTED] wrote:
Original Message
Subject: Re: [Asterisk-Users] Options for 3+ FXO ports
From: "Jorge Mendoza" <[EMAIL PROTECTED]>
Date: Mon, March 08, 2004 8:53 am
To: [EMAIL PROTECTED]
Rich Adamson wrote:
I'm looking into implementing an * solution and I'm expecting to
Tanks
This was exatly what I needed,
/Hans-Henrik Andresen
"Nicolas Gudino" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Hi Hans,
>
> http://bugs.digium.com/bug_view_page.php?bug_id=773
>
> This patch plays a tone 40,30,20 and 10 seconds before absolutetimeout.
_
> Be careful with setting the volume in this manner. This tip might save you
> a bunch of headaches. I adjusted the TX on my X100P cards to low negatives
> values in accordance with the feedback I received from ztmonitor. When I
> achieved the levels that were satisfactory to me an interesting p
> I'm a bit of a linux zero (at least i'm working to know it but it takes
> some time)
> Do i understand :
>
> The call comes in on a card (in this case zaptel) for exampl,e and if in
> the dialplan you have an 'f' extension Hylafax will get the communication
> from the zaptel card ? or is it simpl
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