Almost evey GSM manufactor has these kind of modules.
Ericsson: GM25, DM20,..
Siemens: TC35
(http://www.siemens-mobile.com/cds/frontdoor/0,2241,hq_en_0_2220_rArNrNr
NrN,00.html)
http://www.roundsolutions.com/gsm-modem/index.htm
David
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
C M wrote:
the real
problem is with the asterisk NAT issue. i was asking
for help if any one had similar problem with nufone
account. i am using IAX. is there anything like
nat=yes as in sip.conf?? i read iax should work with
normal configuration. its ok with outbound. i only
have problems with
ok... i tried my * with public ip wioth no firewalls..
seems like its the issue from nuone itself. i'll mail
those guys.
thx.
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
C M wrote:
the real
problem is with the asterisk NAT issue. i was
asking
for help if any one had similar problem
Jared Smith wrote:
acroread is not, but ghostscript and xpdf are ...
On Sat, 2003-11-22 at 21:22, [EMAIL PROTECTED] wrote:
please please please if you are going to write something like that,
write it using something like texinfo or groff or docbook or whatever
so that you can make it available
Serge Mankovski wrote:
Hi
I am trying to dial an extention on my gateway using OH323 without a
gatekeeper.
I would like to be able to do this:
exten=_8.,1Dial(OH323/($EXTEN:1)@xxx.xxx.xxx.xxx,20,r)
You can do it!
Just make sure you are using the latest (1.5.2) OpenH323.
It seems that the only
rocks :)
Alastair Maw wrote:
The Etheral plugin is now actually workable.
A new version is available at:
- http://almaw.com/ethereal-iax2-plugin-0.2.zip
I think some stuff might still be slightly off - unsigned/signed stuff
for timestamps, etc. but it basically works. Expect a pretty much
Michael Manousos wrote:
Serge Mankovski wrote:
Hi
I am trying to dial an extention on my gateway using OH323 without a
gatekeeper.
I would like to be able to do this:
exten=_8.,1Dial(OH323/($EXTEN:1)@xxx.xxx.xxx.xxx,20,r)
You can do it!
Just make sure you are using the latest (1.5.2)
24 2003 10:21 David Luyens :
Almost evey GSM manufactor has these kind of modules.
Ericsson: GM25, DM20,..
Siemens: TC35
(http://www.siemens-mobile.com/cds/frontdoor/0,2241,hq_en_0_2220_rArNrNr
NrN,00.html)
And can it extract from GSM channel GSM encoded voice, just to not making
Does anyone have experience using the Netphone SIP phone from Ortena
Networks (http://www.ortena.com). I contacted them, and they will sell
me 10 units for 95 euros/unit. At least i -looks- better then the
Grandstream :-)
___
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PLEASE DON`T TOP POST
(post reformatted)
On Sat, Jul 13, 2002 at 10:31:53AM -0500, Mark Spencer wrote:
Does anyone (maybe in Europe) know how I could build a GSM compatible
channel for Asterisk, so that one could call other mobile phones from
Asterisk, or build a portable phone system,
On Mon, 24 Nov 2003, Mathew Frank wrote:
http://www.gnokii.org/ comes to mind, though it wont do the voice thing, to
my knowledge.
Maybe bluetooth would be the answer - have the pc register with the phone
under a headset profile, and you'd have your audio. Use AT commands on a
comms profile to
Yes, call control is via serial rs232 and voice is analog interface.
a couple of links where the interfaces are described for the siemens
module:
http://www.cnetek.net/zlxz/Interface_E/TC3x_Interface_v0310.pdf
http://www.conigma.com/downloads/siemens/TC35T/tc35t_hd_01_v0300a_268766
.pdf
David
(Aside 2: Does anyone know why the md driver does not select the fastest
RAID5 checksumming method? I don't use md RAID5 but it always seems to
pick the second-fastest of all the methods it tests, as seen in dmesg.)
IIRC the chosen RAID5 checksum method affect the overall system the
least.
Hi All
Maybe this would be a beter solution, but you may have to buy directly from them
http://www.artech.com.tw/html/gx100e/gx100e.htm
Robb
--- Original Message ---
From: David Luyens [EMAIL PROTECTED]
Sent: Mon, 24 Nov 2003 14:14:10 +0100
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
If you update your source from the CVS, you'll get a new SIP channel
that supports a new syntax for SIP calls in extensions.conf
If you define a SIP peer in sip conf, like
[mysipprovider]
...
You can now use
dial(SIP/mysipprovider/extension)
Where the part mysipprovider is related to the
is it possible to transfer a call while it's ringing
??
SIP/cs001 calls SIP/cs002
The SIP/cs002 user transfers the call to SIP/cs003,
where on SIP/cs003 the phone continues to ring ...
inone way or another (trough manager API or
something else, don't care)
i tried redirect with the
as i said, right now i'm just getting my feet wet. but, i will be needing
to build dialplans on the fly. 'add extension' seems like the right call
to make.
.t
What is the goal of this? It doesn't make much sense to me. Care to
share some insite into what your goal is?
bkw
On Sun, 23 Nov
The Problem: When a call gets into voicemail from Queue and presses 0
before leaving a message * will issue a Hangup. I'm sure it's a context
thing I just don't know where it is. Any suggestions would be appreciated.
Regards, TL
-- Playing 'vm/1/unavail' (language 'en')
-- Hungup
Hi,
-Original Message-
Another issue I've just seen, however :-)
When I'm passing a call from a Zap channel (PRI) I get an
error: STATUS:
Bad
or incomplete voice
This is strange someone else with this issue?
I've also noticed this with iaxComm, so I don't think it is
unsubscribe
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I tried it w/ mine as well and it hung up on me because I just have
Voicemail running not Voicemail2.
It seems as though you have Voicemail2 because it's trying to play the
Unavialable message.
Just a thought though.
Does it do the samething w/
[qout-phillyq]
exten = 0,1,Voicemail(u1)
exten
Hi,
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 5:24 PM
Subject: RE: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
forward - available for download
Hi,
-Original Message-
Another issue I've
Please see the link at the bottom of this and every other email that come from the
list...
- Original Message -
From: Jens Krause [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 10:37 AM
Subject: [Asterisk-Users] unsubscribe
unsubscribe
On Sat, 2003-11-22 at 21:30, [EMAIL PROTECTED] wrote:
On Sat, Nov 22, 2003 at 08:51:35PM +0100, Olle E. Johansson wrote:
But not, alas, in the realm of NAT. Is there any possibility of
removing the broken externip implementation and importing the
patch I submitted that does it properly? If
Hi all!
We set up a sipserver using asteriskX ix66
and need some test calls from around world toverify if it is working
ok.
If you can :-)please call us:
sip:[EMAIL PROTECTED] direct to
snom200
or
sip:[EMAIL PROTECTED] to asterisk
snom200
Thank´s for all
Miklos
iPFONE Telefonia
On Mon, Nov 24, 2003 at 04:59:06PM +0100, Patrick wrote:
On Sat, 2003-11-22 at 21:30, [EMAIL PROTECTED] wrote:
On Sat, Nov 22, 2003 at 08:51:35PM +0100, Olle E. Johansson wrote:
But not, alas, in the realm of NAT. Is there any possibility of
removing the broken externip implementation and
Hello,
I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323.
I tested both oh323 from inaccessnetwork and JerJers chan_h323.
I used 1.12.2 version of oh323 and 1.5.2 version of pwlib.
After latest changes from JerJer chan_h323.c works ok when receiving traffic
from
On Mon, Nov 24, 2003 at 04:59:06PM +0100, Patrick wrote:
Where can I find that patch?
But note that it will not apply cleanly now -- the externip
stuff will conflict. I will try to make a fresh one today.
-w
--
/~\ The ASCII Ribbon Campaign | NO MATTER HOW MUCH DRIVING EXPERIENCE YOU
\
[EMAIL PROTECTED] wrote:
Hello,
I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323.
I would like to add this to the Wiki, but wonder which product you mean in
Cisco's product range?
/O
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On Sun, Nov 02, 2003 at 11:55:12AM -0500, Frank Latini wrote:
unsubscribe
in case you didn't see the footer the first dozen times,
here it is:
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When I have an ip hardphone username setup in my sip.conf :
[109]
type=friend
username=ipphone9
secret=bla-la
host=dynamic
dtmfmode=rfc2833; Choices are inband, rfc2833, or info
defaultip=172.20.0.139
mailbox=109 ; Mailbox for message waiting indicator
callerid=ipphone9
Take a look at:
http://ns1.jnetdns.de/jn/relaunch/asterisk/page15.html
Hope this can help, too...
Samuel
On Fri, 2003-11-21 at 16:22, Cees de Groot wrote:
WipeOut [EMAIL PROTECTED] said:
I would recommend you dump i4l and use a CAPI card with the chan_capi
driver.. The cheap
Hi,
...
So, there's three places this could fail:
1) The frame that's been passed in is empty.
2) The frame is not GSM
3) The frame is GSM, but it's length is not a multiple of 33.
4) The frame is GSM, but could not be decoded.
My guess is that (2) is your problem, although I've seen a
Voicemail1 is gone. Voicemail2 replaced voicemail early this month.
bkw
On Mon, 24 Nov 2003, Tim Thompson wrote:
I tried it w/ mine as well and it hung up on me because I just have
Voicemail running not Voicemail2.
It seems as though you have Voicemail2 because it's trying to play the
All my boxes are working fine with NuFone. You have issues with your
config then.
bkw
On Mon, 24 Nov 2003, C M wrote:
ok... i tried my * with public ip wioth no firewalls..
seems like its the issue from nuone itself. i'll mail
those guys.
thx.
--- Olle E. Johansson [EMAIL PROTECTED]
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls from around
world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] direct to snom200
or
sip:[EMAIL
Hello,
I have a big old Fax machine that will only pick up a ring at 24volts and
20mA(minimum) ring[according to the technical specs manual]. None of my SIP
- Analog phone adapters supply this:
Cisco - 50 volts
SIPURA - 70 volts
Handytone - who knows, but it doesn't work
can anyone tell me if
Tony Kava wrote:
Greetings:
I did some quick searching of my history of this list, and I tried a
quick Google search as well, but perhaps someone on the list can
quickly answer this question. I have a very nicely working Asterisk
system at home with two Digium X100P FXO cards. When my
snippet
; Outbound
exten = _9.,1,Dial(Zap/1/${EXTEN:1},90,Tt)
exten = _9.,2,Macro(fastbusy)
exten = _9.,102,Macro(fastbusy)
/snippet
Use the `group` syntax for the Dial command, ie:
exten = _9.,1,Dial(Zap/g1/${EXTEN:1},90,Tt)
^
And in your
Dial(Zap/g1/
As long as they are in the same group asterisk will pick an unused card.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tony Kava
Sent: Monday, November 24, 2003 12:45 PM
To: '[EMAIL PROTECTED]'
Subject:
unsubscribe
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[EMAIL PROTECTED]
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[EMAIL PROTECTED]
hi.
use groups :)
zapata.conf
group=1
signalling=blah
channel=1-2
etc etc
then in extension.conf, just use
exten = _9.,1,Dial(Zap/g1/${EXTEN:1},90,Tt)
or better, add in globals section
TRUNK=Zap/g1
and
exten = _9.,1,Dial(Zap/${TRUNK}/${EXTEN:1},90,Tt)
in outbound context
matteo
Il lun,
loose username=ipphone9
Not needed.. the [109] is really the username
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 11:42 AM
Subject: [Asterisk-Users] strange SIP authentication/authorization behaviour
When I have an
On 24/11/03 17:51, Bob Knight wrote:
This is way cool stuff. Thanks
Is there any way to put this under the same * cvs control tree?
One stop update.
Well, I'm not sure there's much point. The Ethereal folk are generally
very happy about having yet more packet filters hit their CVS. Once
Setup groups
In your zapata.conf do group=1 before your channels = line.
then Dial(Zap/g1/blah)
bkw
On Mon, 24 Nov 2003, Tony Kava wrote:
Greetings:
I did some quick searching of my history of this list, and I tried a quick
Google search as well, but perhaps someone on the list can
Zap/1 and Zap/2 are analog phone lines. What is the best method of picking
an open line when someone tries to dial-out? i.e. if Zap/1 is in use how can
I instruct Asterisk to use Zap/2 and vice versa? I know complex methods of
making this happen, but I'm sure there is a very simple way to
[EMAIL PROTECTED] wrote:
Although personally I would prefer oh323 for its very well described
config file for now winner is chan_h323.
What is not clear about h323.conf? IMHO, it is a whole lot simpler
than asterisk-oh323's oh323.conf file.
Jeremy McNamara
-Original Message-
From: Tony Kava [mailto:[EMAIL PROTECTED]
Sent: Monday, November 24, 2003 12:45 PM
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] Picking an open channel (FXO port) for
outbound
calls
Greetings:
I did some quick searching of my history of this list, and I
I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
Be brutal. I want to know the gory details, so we can stop any future
pissing matches from even starting by having everything publicly
documented for all newbies.
Jeremy McNamara
Does anyone have experience using the Netphone SIP phone from Ortena
Networks (http://www.ortena.com). I contacted them, and they will sell
me 10 units for 95 euros/unit. At least i -looks- better then the
Grandstream :-)
The phone looks interested and appears to have been on the market for
Walker Haddock wrote:
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls from around world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] direct to snom200
or
On Mon, 2003-11-24 at 10:49, [EMAIL PROTECTED] wrote:
On Sun, Nov 02, 2003 at 11:55:12AM -0500, Frank Latini wrote:
unsubscribe
in case you didn't see the footer the first dozen times,
here it is:
Are you just now catching up on email? Look at the post you replied to,
it is 20 days old.
--
On Mon, 2003-11-24 at 09:12, tad wrote:
as i said, right now i'm just getting my feet wet. but, i will be needing
to build dialplans on the fly. 'add extension' seems like the right call
to make.
If this is so, you may be going about solving your problem completely
wrong. If you are trying to
- Original Message -
From: Conrado Chiappero [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 1:13 PM
Subject: Re: [Asterisk-Users] unsubscribe
unsubscribe
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[EMAIL PROTECTED]
Thanks to everyone for your quick responses to this question. I'm very
excited about the Asterisk project, and the growing community seems to be
very active these days. Hopefully when the time comes for our county's
transition to VoIP we may be able to go for an Asterisk-based solution.
--
Tony
Hi Miklos,
I have the same as Walker.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walker Haddock
Sent: 24 November 2003 18:02
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] test call request
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas
snippet
; Outbound
exten = _9.,1,Dial(Zap/1/${EXTEN:1},90,Tt)
exten = _9.,2,Macro(fastbusy)
exten = _9.,102,Macro(fastbusy)
/snippet
Use the `group` syntax for the Dial command, ie:
exten = _9.,1,Dial(Zap/g1/${EXTEN:1},90,Tt)
^
first you would set up a group in zapata.conf
[channels]
signalling=fxs_ks
group=1
channel=1-2
then in your extensions.conf file replace your dial
line with this:
exten = _9.,1,Dial(Zap/g1/${EXTEN:1},90,Tt)
..Hope this helps...jak
--- Tony Kava [EMAIL PROTECTED] wrote:
Greetings:
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extensions.conf segment:
snippet
; Outbound
exten = _9.,1,Dial(Zap/1/${EXTEN:1},90,Tt)
exten = _9.,1,Dial(Zap/g1/${EXTEN:1},90,Tt) ;search first open zap in
acending orders
or
exten = _9.,1,Dial(Zap/G1/${EXTEN:1},90,Tt) ;search first open zap in
decending order
asuming in
I am trying to get the following phones for testing. Is there a distributor in the US
that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco
phones there too hard to configure and too expensive!
1 - Sipura SPA-2000
2 - Grandstream Sip phone BT-102
1 - Grandstream
Hi !
Thank you for the call
I think that you have to Put reinvite=no in your sip.conf for the given
friend/user/peer to keep * from trying a native bridge.
I tryed to call you ( sip:[EMAIL PROTECTED] and
sip:[EMAIL PROTECTED]) but the call timeout
Thank you again
Miklos
- Original
Adding canreinvite=no to your sip.conf for that phone should do it..
I did have that in there. Here's the stanza in sip.conf. I set up device [90]:
[90]
context=from-sip
type=friend
insecure=yes
host=sipserver.com.br
fromdomain=home.datacrest.com
canreinvite=no
reinvite=no
nat=yes
Does * supports time zone setting per EACH user/device?
Ta
SJ
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Hello Jeremy,
we are using asterisk for some of our services long time ago especially
SIP and H323 channels - around 10 000 - 15 000 minutes per day.
Regarding oh323 and h323 channels I have to say my opinion is that h323
channel have much better support for some exotic codecs as g72. than
On Mon, Nov 24, 2003 at 12:40:00PM -0600, Steven Critchfield wrote:
Are you just now catching up on email? Look at the post you replied to,
it is 20 days old.
Wow, you're right. I have noticed some delays in posting to the
list, but 20 days is a bit excessive. But, honest, it just landed
in
Hi,
It is possible to place an IAX phone in the same pickupgroup (1) as the 4
phones connected to a TDM400 card?
I have tried to put
pickupgroup=1
in the iax.conf general section too, but I cannot pick an internal call
using *8 from the IAX phone.
There is any other way to obtain this
Why do we even bother to try and point these people in the right
direction?
I agree. A trick I've used in the past is to filter the word 'unsubscribe'
to incoming email and using some other metrics (mostly quantity of body
text), unsubscribe them automatically. It's just not worth fighting
Hello,
Ok, I've tried tinkering with the ring voltage (this is an editable variable
on the Sipure SPA-2000) and I still can't get it to work. I plug a generic
POTS line in and it works, I plug an analog port from our old Comdial PBX in
and it works.
with the sipura adapter I've tried changing
Ok... I know that ya'll like to point people to the web page listed at the
bottom, but how many of ya'll have put on your stuipd cap and looked at the
page? It does not readily direct you to unsubscribe. Now if you stop and
take off your stupid cap and read the whole page you will see at the
Hi dave
I think that is a problem with nat, calls direct to the snom phone trough
ix66 works well but from asterisk don´t.
Thanks for the call
Miklos
- Original Message -
From: David J Carter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 5:04 PM
Subject:
Please add canreinvite=yes and, when the * is behind a NAT router,
nat=yes to section [nikotel]. As a nikotel customer, you can also
open a ticket and request help from nikotel.
Michael
Daniel Chabrol wrote:
Hello list!
I'm using Asterisk CVS-11/22/03-04:28:51 and try to route my normal
Here is a little bit harder question:
I want my local outbound calls to use an FXO interface
as described in this thread however...
If there is no available FXO interface then I'd like the
called go through my SIP service provider who will gateway
the call back to PSTN for me (for a small per
The subject pretty much says it all. I have a customer who would like
to have an option where a caller can leave a voicemail in such a fashion
that it would be simultaneously delivered to a set of mailboxes all at
once--the idea is trouble ticket type operation where multiple
technicians will
I am trying to get the following phones for testing. Is there a distributor in the
US that is
able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones
there too
hard to configure and too expensive!
1 - Sipura SPA-2000
2 - Grandstream Sip phone BT-102
1 -
I'm using Voicemail2. Either way my systems issues the hang up w/v1 or v2.
TL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday, November 24, 2003 12:54 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pressing 0 in Voicemail
The Etheral plugin is now actually workable.
A new version is available at:
- http://almaw.com/ethereal-iax2-plugin-0.2.zip
I think some stuff might still be slightly off - unsigned/signed stuff
for timestamps, etc. but it basically works. Expect a pretty much final
v0.3 some time soon that
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 2:46 PM
Subject: Re: [Asterisk-Users] test call request
Adding canreinvite=no to your sip.conf for that phone should do it..
I did have that in there. Here's the
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Florian Overkamp
Sent: Monday, November 24, 2003 11:58 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
forward - available for download
On Mon, 2003-11-24 at 13:56, [EMAIL PROTECTED] wrote:
On Mon, Nov 24, 2003 at 12:40:00PM -0600, Steven Critchfield wrote:
Are you just now catching up on email? Look at the post you replied to,
it is 20 days old.
Wow, you're right. I have noticed some delays in posting to the
list,
Or it could be worse still
They could be you
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, 25 November 2003 6:39 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] unsubscribe
Why do we
Do these analog - SIP VOIP adapters truely supply the 24V that a POTS
line
supplies?
Strictly speaking, the telephone line isn't 24V.
it's a 20mA DC current loop, with a ~80-110VAC ring signal.
Anybody have any ideas why this fax machine won't work with any analog
adapter I've tried?
It's
I know you'll never entirely be rid of these types of requests on
any list, but some quick usability testing might help reduce the
total number of them. :)
Some observations:
- To a user who is unfamiliar with listserv software (or one
who is used to older ones that take commands on the
Ok... I know that ya'll like to point people to the web page listed at
the
bottom, but how many of ya'll have put on your stuipd cap and
looked at the page? It does not readily direct you to unsubscribe. Now
if you stop and take off your stupid cap and read the whole page you will
see at
On Monday 24 November 2003 12:27, Andrew Nelson wrote:
WOW!! It took almost 1 hour before my post made it back to me.
-Andrew
Ok... I know that ya'll like to point people to the web page listed at the
bottom, but how many of ya'll have put on your stuipd cap and looked at the
page? It
So the generized question is how to do across channel type
fail overs?
From a post to this very thread just a few hours ago:
exten = _9.,1,ChanIsAvail(Zap/1Zap/2)
exten = _9.,2,Dial(${AVAILCHAN})
exten = _9.,102,NoOp
exten = _9.,103,Congestion
Now in your case, you'd say
exten =
Anybody have any ideas why this fax machine won't work with any analog
adapter I've tried?
Have you verified you have the right tip-ring polarity? Maybe this is one of
those few devices that it makes a difference when it is backwards.
___
mail alias on the mailserver?
Matteo
Il lun, 2003-11-24 alle 21:33, Brian Capouch ha scritto:
The subject pretty much says it all. I have a customer who would like
to have an option where a caller can leave a voicemail in such a fashion
that it would be simultaneously delivered to a set of
I assume that you have tested a analog phone with your ATA and it works.
If this is true, I suspect that your fax machine does not work with 24
VDC but 48 VDC. POTS line and old PBX supply -48 VDC.
Jorge
mattf wrote:
Hello,
Ok, I've tried tinkering with the ring voltage (this is an editable
Quoting Olle E. Johansson:
I managed to terminate calls from cisco: as5300 and 7206 to asterisk over
h323.
I would like to add this to the Wiki, but wonder which product you mean in
Cisco's product range?
AS5300 and 7206VXR equipped in apropriate voice processing cards are high
density voip
Brian West wrote:
Ya learn to search the archives. This has been covered MANY MANY times.
And I still haven't gotten the echo to go away completely. I usually
just end up making it worse. :)
bkw
On Sun, 23 Nov 2003, VoIP Fan wrote:
Hello:
I have installed *. I configured my SIP
I actually was interested in the same question and this is the answer I was
looking for. By looking at the original question I think he is having the same
issue I am having here. I have two FXO lines coming in and I have a two line
FXS phone as well. When you pick up line 1 FXS I want it to try
On Mon, 2003-11-24 at 14:33, Brian Capouch wrote:
The subject pretty much says it all. I have a customer who would like
to have an option where a caller can leave a voicemail in such a fashion
that it would be simultaneously delivered to a set of mailboxes all at
once--the idea is trouble
Hi ho!
Does anyone have experience using the Netphone SIP phone from Ortena
Networks (http://www.ortena.com). I contacted them, and they will sell
me 10 units for 95 euros/unit. At least i -looks- better then the
Grandstream :-)
The phone looks interested and appears to have been on
Lubomir Christov wrote:
BUT I have the say that I have the same opinion as martin
([EMAIL PROTECTED]): Although personally I would prefer oh323 for
its very well described config file for now winner is chan_h323
Again, what is not clear about h323.conf? It follows the other Asterisk
channel
From: Jeremy McNamara [EMAIL PROTECTED]
I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
Be brutal. I want to know the gory details, so we can stop any future
pissing matches from even starting by having everything publicly
I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
For 6 months, I didn't know what was a perfect connection using
both except using G711 with oh323. Few weeks ago, a big mind
from Australia solved the problem with 1 or 2 lines of
These instructions are mostly good, I have some corrections below...
On Thu, Nov 20, 2003 at 09:14:24AM +1100, Anthony Wood wrote:
Hi * Fans,
I have some fritz cards now, followed instructions from stuart hirsts email of Jun
28:
-
Thanks for your info but I think I have it working at
How about using symlinks for the mailbox directories?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo
Sent: Monday, November 24, 2003 4:44 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] One voicemail -
Hi,
I'm after some ISDN card resellers in Australia. I've noticed the AVM cards
should work well here, however I've found the reseller here has no website,
is listed in Yellow pages as management consultants (??) and when I called
quoted almost $400.
Does anyone know where I can get the AVMs in
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