PMS is the correct term for the hotel billing systems. Property Management
System. Problem is that they are all proprietary interfaces and it is very
hard to get the major companies to work with you. I've done so in the past.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Any network device (ie: switch, router, firewall) will add a small amount of
latency. To test the latency your firewall adds, you could simply try to do
a ping www.google.com, directly in front and behind the firewall, and look
at the ms response times.
Cheers,
S.
I've noticed with the CVS build I have, if I restart * it shuts down and
then restarts and then shuts down and then restarts until I reboot my
system.
Anyone else have this problem?
S.
smime.p7s
Description: S/MIME cryptographic signature
___
Me neither.. but just started receiving now. WEIRD.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Monday, August 01, 2005 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] test message -
I have no spam lists. :P
It died for many people I know.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, August 01, 2005 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] what is
Check out http://aussievoip.com.au/wiki-G723-1-Install
How to G723 for *
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Apu Islam
Sent: Wednesday, July 27, 2005 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Try a -D 2005-05-29 16:47
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick
Baranowski
Sent: Thursday, July 28, 2005 12:10 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Get older CVS version
Well
IDK if this might help :-)
http://www.voiptroubleshooter.com/diagnosis/usersymptoms.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jochen Witte
Sent: Thursday, July 28, 2005 2:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Check out: http://www.voip-info.org/wiki-VOIP+Payphones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Thursday, July 28, 2005 4:03 AM
To: Asterisk-Users
Subject: [Asterisk-Users] Public phone
A client wants to put phones in
I know this isn't directly related to * but I found it works very well in my
voip environment. Check out cFosSpeed @ www.cfos.de. It gives you QoS based
on applications and also seems to have increased my network throughput.
Cheers,
S.
smime.p7s
Description: S/MIME cryptographic signature
I dont know if I have the same experiences.
Usually my Skype calls are very garbled at first. I find that my G729 Asterisk
calls are better quality. You can try using ULAW if you have the
bandwidth. It. might make the quality sound better.
Maybe its your SIP client/hardware
phone that
You might want to try this group out: http://groups.yahoo.com/group/pa1688/
Most of these Chinese phones are using the pa1688.
Cheers,
Storm.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong
Sent: Friday, July 15, 2005 12:34 AM
To: Asterisk
I found the problem was with eyeBeam when I had more than one video codec
enabled. Try on eyebeam to only have h263p enabled.
Does the video appear in the Echo test?
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald_Wiplinger
Sent: Monday,
I believe you can find it here:
http://bugs.digium.com/bug_view_page.php?bug_id=759
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kamran Ahmad
Sent: Sunday, July 10, 2005 10:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how
Are you sure that the video is set up correctly? If you have a cheap webcam
you have to turn off video hardware acceleration.
Cheers.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Blake Krone
Sent: Friday, July 08, 2005 5:53 AM
To: Matt Riddell
Title: Asterisk/Grandstream Budgetone disconnect issue
Might want to try updating your firmware
on the Grandstream. I use this version, works great: http://gs-firmware.gratissip.dk/firmwares/1.0.6.7/
Cheers,
S.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of
Bruce,
I too am interested in the telephone number for SimpleTelecom, as my company
had put quite a large prepayment to them. You said you posted the number on
this list; I searched for all post by you and did not find the posting which
contained a phone number. Would you be so kind as to
Hi,
Today I decided to upgrade my * PBX and compiled the latest Development Head
and installed it. I keep getting this message:
WARNING[18268]: loader.c:313 __load_resource: libspeex.so.1: cannot open
shared object file: No such file or directory
Jul 5 01:26:32 WARNING[18268]: loader.c:523
I never saw much on API info; maybe it's out there somewhere? I had to hack
around. It starts to make sense fast.
Start by taking a look at ./asterisk/apps/app_skel.c and then look at other
apps and you can start to figure out how they do things fast.
Good luck,
S.
-Original Message-
Said all that, I still would love to have a Skype channel for *. Hopefully
they will release a Linux API so people can start to play (including me).
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Saturday, March 26, 2005 12:50 AM
Hi,
I use the windows client SJPhone (http://www.sjlabs.com) to make direct SIP
calls. To call my * box I just enter: sip:[EMAIL PROTECTED]
SIP:10.100.0.201 to call my SIP handset.
Good luck,
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of CuPoTKa
I've used sipphone.com before. Their rates are not the greatest.
S.
-Original Message-
why nobody use sipphone.com to connect to asterisk ?
Best Regards
Zhao Zigang
Alcatel Shanghai Bell Co., LTD
*:388,NingQiao Rd.,Shanghai 201206
*:086-21-50554550-7762
*:[EMAIL PROTECTED]
I've successfully implemented several VOIP over WiFi networks in the UK with
excellent quality. I am currently in Canada.
The trick is designing a good wifi backhaul network in quasi full duplex.
Storm.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Are there any VOIP lobbyist groups in Canada?
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon
Patterson
Sent: Monday, January 17, 2005 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Canadian
I put .20 on my tftp server and the BT100's uploaded fine. Before I was
using .16 because .18 had some issues. We have been running .20 for a month
now and no problems reported.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of My Other Email
Sent:
This may be a bit off topic, but here is the best Speech recognition I have
seen so far:
VLVR -- Very Large Vocabulary Speech Recognition
http://akpublic.research.att.com/projects/mohri/vlvr/
Cheers,
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
What if you call an external system and get a voicemail. Press # to finish
your message . you would have to press ##.
IMHO I think most users are not sophisticated enough to transfer calls. If
they are they can press ##.
Or am I missing something? :)
S.
-Original Message-
From:
to transfer function. Get
# as first character in transfer, send out the DTMF tones instead and drop
the request to transfer.
I could be all wet on this, but my feeble mind sezs this makes sense from a
programming perspective.
Lyle
- Original Message -
From: Storm D. J. Petersen [EMAIL
Ive had
similar problems with grandstream phones.
They lock up for no apparent reason and after that no matter what you do
they are dead. I must of sent back
15 so far like this.
S.
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Title: SIP phones
Why dont
you use an ATA device with a loud regular phone and/or hook up one of those
really loud ringing devices you can get at a phone shop? J
Just a
suggestion.
S.
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Which Sat company are you using?
I struggled and got it working on Aramiska but their latency is up to 4.5
seconds. Haven't figured out why it's so long but assume they have some bad
routing issues - as they are now going to offer a new service that
guarantees 600ms.
S.
-Original
Hi,
I'm looking for a database with all the world's country codes and area
codes. Can anyone point me into the right direction?
Cheers,
S.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
.internal.petercox.aramiska.net
[192.168.1.4]) by ip-10-2-82-148.arc.aramiska.net (Postfix) with
SMTP
id 67CE64A; Thu, 2 Sep 2004 10:05:03 + (UTC)
From: Storm D. J. Petersen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users
Hello,
I have a problem with jitter over a 2mb up 1mb down satellite connection. I
call my friend over the satellite - I call perfect but they cannot make out
a word I say. However if I leave him voicemail on his asterisk box, it
records my voice perfect. I have this problem when calling other
, it
records my voice perfect. I have this problem when calling other people as
well.
Storm D. J. Petersen
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J
I don't mind latency ... it's the garbage jitter where no one can understand
a word.
Interestingly enough if I do this it works fine:
[grandstream 1]- [sat]- [pbx in mothers house]
[grandstream 2]- [sat] -/
where the grandstream phones are side by side.
S.
-Original Message-
From:
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I get the
following message:
Chan_iax2.c:5251 socket_read: Reject connect
Does anyone have any interesting SIP service numbers like FWD 411 Tell-Me ?
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jay Milk
Sent: Monday, June 14, 2004 4:54 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] where can I get toll-free number?
I got the CVR last night and compiled it. I didn't get any problems
compiling. But now my uniquieid is not being populated.
Any ideas?
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield
Sent: Sunday, June 13, 2004 12:47 AM
To: [EMAIL
Whoops I didn't RFM... :( I forgot the CFLAGS+-DMYSQL_LOGUNIQUEID
Sorry.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield
Sent: Sunday, June 13, 2004 12:47 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: cdr_addon_mysql.c
In
Hi,
Just a
note to whoever cares that the CVS astman application does not seem to
compile. I copied the /usr/src/asterisk/astman/Makefile
into the CVS astman directory and it compiled and ran fine.
S.
PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: 11 June 2004 00:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I found that the PREPAID system didn't disconnect proper and tracked
time from when you dialed not when your phone made connection. I
Hi,
Don't try to use my patch with the latest app_dial. It will only work with
Release 1.0. Mine is just very clean and simple implementation to force a
disconnect X seconds after a call was bridged.
I was skimming though the latest source tonight and you are right, lots of
nice features.
I
Hi.
Thanks for tipping me off with the new firmware. I installed it and tested
the codec. Has more delay but seems to be better quality than what I was
using before.
Anyways, that didn't fix the SIP Registration Failure that I am getting.
Any ideas?
S.
-Original Message-
From: [EMAIL
Hi, I found that the PREPAID system didn't disconnect proper and tracked
time from when you dialed not when your phone made connection. I ended up
making my own system and had to modify the Dial app.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Umar
Hi,
Can someone explain (or give me a link) in easy and no uncertain terms what
the deal is with MySQL and Asterisk. If I run MySQL 3.x? Do I have to have
a license if I sell my products? What if I sell a server with Asterisk on
it?
Thanks,
S.
___
] Behalf Of Storm D. J.
Petersen
Sent: 11 June 2004 00:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I found that the PREPAID system didn't disconnect proper and tracked
time from when you dialed not when your phone made connection. I ended up
making
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: 11 June 2004 00:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I found that the PREPAID system didn't disconnect proper and tracked
time from when you
Thanks!
Why did * Pull the MySQL support in the current version?
Storm.
Storm D. J. Petersen
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, June 10, 2004 8:02 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] GNU Licenses, Asterick, MySQL, andtheUniverse?
On Thu, 2004-06-10 at 21:49, Storm D. J. Petersen
the call was bridged (connected) - as
apposed to disconnecting in x seconds after the call was dialed.
Feel free to ask questions.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: Thursday, June 10, 2004 7:48 PM
To: Asterisk-Users
I haven't looked at the CVS source yet - I will take a look and see if it's
the similar or different.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Murphy
Sent: Thursday, June 10, 2004 9:30 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
into the CVS now. ^_^
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: Thursday, June 10, 2004 9:45 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I haven't looked at the CVS source yet - I
Hi,
I have an * server on a routable (public) IP address and a sip client behind
NAT using a Grandstream phone. He is connected through a bi-directional
satellite so he has a bit of latency involved. Usually I can dial this
extension and them to me. But I keep getting a registration failed
Hi,
I'm new to asterisk. After fiddling a bit I got it to work. It seems great.
One question though, is it possible to configure asterisk when it is
running? i.e. add new phones or do you have to restart it every time you
want to make changes?
Thanks,
Storm.
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