-Original Message-
Anyone here with experience on the Cisco ATA 188 and *?
Is it as good as ATA 186?
AFAIK the only difference was the built-in switch so you can plug your PC in
the back, right ? Should be 'as good'.
Florian
___
What ver of SJPHONE?
Thanks for the voicemail stuff :-)
- Original Message -
From: Girish Gopinath [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, February 28, 2004 7:48 PM
Subject: RE: [Asterisk-Users] DTMF Issues with SJPHONE
Has anyone had a similar issue with Asterisk
Anyone got an example of sip and extensions confs
for Iconnect outgoing calls behind NAT.
What ver of SJPHONE?
SJPhone Evaluation Version, release Jul 31, 2003, Build: 1.10.187c
Girish
_
All the news that matters. All the gossip from home.
http://www.msn.co.in/NRI/ Specially for NRIs!
On Fri, 2004-02-27 at 21:39, Chad Sawyer wrote:
In the contrib/scripts directory I have been trying to figure out the
format of the entries in the MySQL table.
-CUT-
There are 3 different approaches to storing users in a database.
The first is dynamic - the user details are read directly from
Dear All,
Thanks, but I am not using chan_oh323, I am using chan_h323. The major
reason why I am not using chan_oh323 is because of a bug that Michael is not
yet able to resolve. Every call that goes out via this h323 channel will be
considered connected and picked up (false answer supervision)
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David Hajek wrote:
| Is there english version of their sipgate.de website?
no ... I just tried the google translater - it did not work (for me) I
think the translation programs don't work with php pages...
Birk
|
| -D
|
|
|-Original Message-
Hi,
Are you behind a NAT/Firewall?
dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004 11:04
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
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The Server I use is somewhere in the Internet with a real ip. Myself and
others connect to the server via vpn in order to go through various
firewalls. Since I can get calls but only can't place calls (via
sipgate.de) I don't think it is a firewall
Hi again,
What is your sipgate number, I have just setup my asterisk to call a sipgate
numbar and it rings.
If you want to call me, then try my IAXTEL # 1 700 818 8820
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004
Birk,
Even using VPN to get to the server you will still have I assume a private
IP address on the VPN side. This will pass through a NAT/Firewall to the
outside world. This may or may not be on the server you connect to, but I
would bet you still pass through a NAT/Firewall.
I assume your
FWIW... Version 6.2 of Cisco's sip code for the 7940/7960 was posted on
Cisco's download site Feb 17th.
The v6.2 release notes suggest the following caveats were addressed:
SIPPhone: CANCEL messages not formatted properly after 180 received
Branch ID is not compliant to RFC3261
SIP IP Phone
I wonder if the next is possible
with *:
PABX
|
E1
|
PABX E1- Asterisk
E1 PABX
| \
E1 \
| IP
PABX \
Cisco
827V
Analogue PBX
If possible, how much power
the CPU must have?
Much appreciate any help.
Nikolay Koev
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 29, 2004 1:44 AM
Subject: OTish: Firefly Carshing with *
Firefly seems to be crashing when I dial from the console (i.e. Dial
[EMAIL PROTECTED]) but works fine from telephone...
Also, it cuts
Hi,
-Original Message-
Firefly seems to be crashing when I dial from the console (i.e. Dial
[EMAIL PROTECTED]) but works fine from telephone...
Also, it cuts my ringing sound off after about .5 seconds.
What version are you using ? There was a small bug in Firefly, fixed last
carl wrote:
Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT.
Here you go:
[Scene starts out with you on the phone with IConnect technical support.]
You: I know that Asterisk isn't one of your supported platforms. I'm not
asking you to support my
Nikolay Koev wrote:
I wonder if the next is possible with *:
PABX
|
E1
|
PABX E1- Asterisk E1PABX
I am look into Galaxy Voice's service as they provide numbers in my
area. Has anyone on the list had any experience using this service with
Asterisk?
I am interested to know if it can be made with work with Asterisk.
How is the quality of sound? Are there limits with the number of
If I want users to be able to call each other (or others to be able to
call users on our Asterisk system) using their email address
([EMAIL PROTECTED]) what would have to be done?
I am guessing the folowing..
In sip.conf the phone definition would have to be..
[user.name]
secret..
blah..
In
I am new to * and I am trying to set up a test box with a ISDN card with the
cologne chip set (twin towers on the isdn chip) . I have downloaded
bri-stuff.0.0.2.rc12 from www.junghanns.net site. I would like to test with
a driver that supports echo cancellation in software.
I am running the
That's all you need. At least, that's kinda how I have mine set up and it
works fine to dial-by-email.
WipeOut wrote:
If I want users to be able to call each other (or others to be able to
call users on our Asterisk system) using their email address
([EMAIL PROTECTED]) what would have to be
On Fri, 2004-02-27 at 21:39, Chad Sawyer wrote:
In the contrib/scripts directory I have been trying to figure out the
format of the entries in the MySQL table.
It isn't at all obvious is it?
I've now worked out what it does have written this up on the Wiki,
along with my previous post about
I am a new asterisk user. I have had a box up and running for a couple
of months and been very happy with it. Last night I came up with a
question that I have not been able to find an answer too. I purchased 5
licenses for the G729 codec from digium. My source is current from CVS
as of
Matt wrote:
I looked at my sample config's and I cannot find an example of an
extension where you specify the codec differently for each extension.
Can someone show me a sample extension?
You're looking in the wrong place. (I should have been more specific.)
You specify the codecs when you
I forgot to mention what I have been trying to fix it. I'm running it
from the console asterisk -vvvcng but this does not help. I've
searched the mailing lists and found a lot of messages with people
having the same problem. I'll try calling digium Monday if I cannot
resolve it today and
NOt sure if there is an official download site, but I just recieved a copy of
the updated firmware from pulver. I can send it to you if you like. I have
emailed back asking for instructions on how to load.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax
Hi Johnathan,
I wouldn't mind a copy of the firmware if you could send it.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jonathan
Moore
Sent: 28 February 2004 19:24
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] wisip firmware,
The original message isn't copied because its HTML encoded.
Basicly the author wanted to know if he could use a HT-286 (Grandstream)
to bypass the PBX, generate busy, answer calls, etc.
The Grandstream HT-286, and the Sipura SPA-2000 are both
ATA FXS based devices.
ATA == Analog Telephone
From a New Mexico perspective,
When you order a PRI from a CLEC they typically will dump
your CLID info and replace it with the main number on the
span. You can request that they not do this and that they
pass the CLID thru.
We have been working with some of the local PSAP's here,
CLEC's and
Hi all,
I am currently testing Klaus-Peter Junghanns' zaphfc bri driver
0.0.2rc12 with two HFC ISDN cards in PtP setup - one connected to telco,
one to the legacy pbx - and try to dial from a pbx extension out to the
pstn through astersik.
This works perfectly as long as I dial on hook and
Same as mine. Strange!
I'll keep trying. Cheers.
- Original Message -
From: Girish Gopinath [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, February 28, 2004 9:53 PM
Subject: Re: [Asterisk-Users] DTMF Issues with SJPHONE
What ver of SJPHONE?
SJPhone Evaluation Version,
I'm considering a small office setup with at least 12 extensions.
Seems (as has been stated in previous threads) that for the FXS ports, a
T100P and a channel bank could be the most cost-effective way to do this.
I've got * set up w/ one X100P and one TDM400P, and have been very happy
with it. I
I signed up with nufone. Their customer service is a little bit slow
but they seem to be pretty decent. I'd recommend checking them out.
www.nufone.net
Darren Wiebe
[EMAIL PROTECTED]
Carl wrote:
Ha ha I get the picture :-)
I've tried Voicepulse but can't manage to get through with them
Hi,
Can someone offer some assistance in setting up
Asterisk as a gateway to connect to a third party gatekeeper.
I have looked at the h323.conf.sample file but not
sure of the following:
Do I need to create a new h323.conf file?
Where should this file reside i.e., h323
directory?
I'll give them a whirl. Cheers C.
- Original Message -
From: Darren Wiebe [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 29, 2004 11:00 AM
Subject: Re: [Asterisk-Users] Iconnect behind NAT
I signed up with nufone. Their customer service is a little bit slow
but they
Carl wrote:
I'll give them a whirl. Cheers C.
Carl, are you not getting my emails?
John
___
Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Carl wrote:
I'll give them a whirl. Cheers C.
If you email me a username/PW combo, I'll get you an account set up and
email you the particulars for this side (or telephone you if you include a
number) as soon as I get home from dinner.
John
___
There are new iaxComm binaries for Windows, Linux and Mac OSX posted at
http://iaxclient.sourceforge.net/iaxcomm/index.html
These binaries also have the recent library change that allows client to client
connections to be handed off correctly.
Recent changes include speakerphone mode, blind
how is your outgoing dialplan?
tried into specifing something like
exten = _XXX.,1,Dial(blah/${EXTEN})
note the point : this rule will match
at least 4 digits, but also 5,6,7...N
matteo
Il sab, 2004-02-28 alle 22:34, Jan Baumann ha scritto:
Hi all,
I am currently testing Klaus-Peter
Hello all,
trying the zaphfc driver from Klaus-Peter Junghanns with Cologne-Chip
PCI card I experience a clearly audible crackling sound during calls
through the Zap BRI channel to PSTN. Calling the same destination from
the same SIP extension via sipgate.de the sound is perfect.
What I hear
pcphoneline.com sells a little box with two RJ-11 jacks that is supposed to
convert an FXS port into an FXO port. According to their blurb, when a call
comes in it basically conferences the two lines together. Is anyone out
there using this box with Asterisk? Any problems?
What happens to
Anybody know how to implement a hotel wake-up call feature with *?
--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with my spam filter!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Sat, Feb 28, 2004 at 08:39:26PM -0500, Bill Michaelson wrote:
Anybody know how to implement a hotel wake-up call feature with *?
I just wrote an AGI for it. I literally just got it working the day
before yesterday, so it's not really 'pretty' yet. I also don't have
all of the voice prompts
Bill, tell us about your system!
How many rooms, what kind of extension set in the rooms, number of
outside lines, front desk capabilities, how you bill back tel charges
to the room, etc.
Have you worked-out the ratio of guests to outside lines? IVR? Do you
use the directory function for
Arretni VoIP Tech wrote:
hello,
Is it normal that * starts its billing when voicemail starts to
prompt? can I do something like it will only start to bill if the
caller left a message? right now, im seeing that unanswered calls
that are forwared to voicemail are considered billable as well
I'm trying to play around with ENUM, and John Todd helped me last night
on the IRC channel in terms of finding this site and other docs to get
me going.
But now I wonder: how can I test it? I started by trying to randomly
try every possible number in the US, but soon tired of that approach. .
Brian Capouch wrote:
I'm trying to play around with ENUM, and John Todd helped me last
night
on the IRC channel in terms of finding this site and other docs to get
me going.
But now I wonder: how can I test it? I started by trying to randomly
try every possible number in the US, but soon
Title: If one extension is busy...
One of my users has two extensions, both of which ring simultaneously when a call comes in for her. This works fine.
If her primary extension is busy, then she is on the phone and there is no reason to ring the secondary extension. In this case, the call
On Sat, 2004-02-28 at 19:39, Bill Michaelson wrote:
Anybody know how to implement a hotel wake-up call feature with *?
You could modify my callback script. It would require some pretty
significant changes, but it's a good place to start. You can find it,
and other scripts, (some good, some
Bill Michaelson wrote:
Anybody know how to implement a hotel wake-up call feature with *?
It seems like it could be accomplished with an AGI and a script that
wrote call files. Have the AGI prompt for the wakeup time (or have a
web interface for a front-desk person do it) and write a file to a
Hi Floks,
I am just starting with * and while playing with the demo configuation I
notice that the CPU utilization is 98-100% no matter if I am leavin a
message or listening to the various voice prompts. Is this normal?
The system is a P4 1.6GHz / 512MB running redhat 7.3 and kernel 2.4.22
I would be interested in the AGI Script. As for the voice prompts, I
am having Allison record some stuff for me on Monday, including prompts
for such a wake up system, that I plan to donate back to the Asterisk
community.
This is what I have for Allison:
Wake up call! This is
Hello,
Pls. help !
I have server on Freebsd 5.2 and don't may install
asterisk , following errors: ( gmake clean ; gmake install )
-
iasing ruleshash/ndbm.c: In function
`dbm_store':hash/ndbm.c:185: warning: dereferencing type-punned
All the digits should already be recorded so you could easily skip that
part and play back any digit from the AGI 1-9 that it was assigned.
On Sun, 2004-02-29 at 00:03, Robert Lawrence wrote:
I would be interested in the AGI Script. As for the voice prompts, I
am having Allison record some
You know what would be nice?
If Firefly could have a Network to use assigned to a contact.
I.E. I use 800 to check my voicemail at work and call work extensions etc so
I have to have IAX as my internal calls...but this means I can't contact
people on the firefly network...
Kind regards,
Matt
if u add #'s to your contact list w/ @networknameinyourclient
they are connected thru that network such as firefly or others
On Sun, 2004-02-29 at 15:05, asdasd wrote:
You know what would be nice?
If Firefly could have a Network to use assigned to a contact.
I.E. I use 800 to check my
sweet, cheers
- Original Message -
From: William Suffill [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 29, 2004 8:44 PM
Subject: Re: [Asterisk-Users] OTish: Firefly Crashing with *
if u add #'s to your contact list w/ @networknameinyourclient
they are connected thru
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